File size: 21,034 Bytes
93533d4
10a7d9a
93533d4
 
 
 
 
dff3cf1
 
fd13715
6645145
93533d4
fd13715
202a8b5
fd13715
feaa89d
fd13715
 
 
 
 
feaa89d
fd13715
feaa89d
 
 
93533d4
 
c102c06
202a8b5
c102c06
 
 
93533d4
 
 
 
 
 
 
 
61e51f4
 
93533d4
 
 
 
 
61e51f4
93533d4
61e51f4
93533d4
4d24677
 
93533d4
 
 
 
61e51f4
 
 
93533d4
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
61e51f4
 
 
 
 
 
 
 
 
 
 
 
93533d4
 
 
 
 
 
 
c102c06
 
 
 
4d24677
 
e4effb9
4d24677
 
 
c102c06
4d24677
 
93533d4
 
 
 
 
 
 
 
c102c06
e4effb9
 
 
93533d4
c102c06
93533d4
c102c06
4d24677
 
e4effb9
 
93533d4
 
 
3141b41
e4effb9
93533d4
e4effb9
 
 
93533d4
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
e4effb9
 
 
 
 
 
 
 
 
 
93533d4
 
e4effb9
 
93533d4
c102c06
93533d4
 
 
 
 
 
 
 
 
c102c06
93533d4
e4effb9
 
93533d4
c102c06
93533d4
c102c06
4d24677
93533d4
 
4d24677
93533d4
 
 
e4effb9
 
93533d4
e4effb9
 
 
93533d4
 
 
 
 
 
 
 
 
 
4d24677
93533d4
 
 
 
 
 
4d24677
 
 
93533d4
4d24677
 
 
 
 
 
e4effb9
4d24677
e4effb9
 
 
 
 
 
 
 
4d24677
 
e4effb9
4d24677
 
e4effb9
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
4d24677
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
e4effb9
 
 
 
 
 
 
 
4d24677
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
93533d4
 
 
 
 
 
 
e4effb9
 
 
 
 
 
 
 
93533d4
 
4d24677
 
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
---
language: zh-TW
datasets:
- common_voice
tags:
- audio
- automatic-speech-recognition
- hf-asr-leaderboard
- robust-speech-event
- speech
- xlsr-fine-tuning-week
license: apache-2.0
model-index:
- name: XLSR Wav2Vec2 Taiwanese Mandarin(zh-tw) by Voidful
  results:
  - task:
      name: Speech Recognition
      type: automatic-speech-recognition
    dataset:
      name: Common Voice zh-TW
      type: common_voice
      args: zh-TW
    metrics:
    - name: Test CER
      type: cer
      value: 18.36
---

# Wav2Vec2-Large-XLSR-53-tw-gpt
Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on zh-tw using the [Common Voice](https://huggingface.co/datasets/common_voice).   
When using this model, make sure that your speech input is sampled at 16kHz.

## Usage
[Colab trial](https://colab.research.google.com/drive/1e_z5jQHYbO2YKEaUgzb1ww1WwiAyydAj?usp=sharing)

```
import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
    AutoTokenizer, 
    AutoModelWithLMHead 
)
import torch
import re
import sys

model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"

chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\"#$%&()*+,\-.\:;<=>?@\[\]\\\/^_`{|}~]"


model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

tokenizer = AutoTokenizer.from_pretrained("ckiplab/gpt2-base-chinese")  
gpt_model = AutoModelWithLMHead.from_pretrained("ckiplab/gpt2-base-chinese").to(device)

resampler = torchaudio.transforms.Resample(orig_freq=48_000, new_freq=16_000)

def load_file_to_data(file):
    batch = {}
    speech, _ = torchaudio.load(file)
    batch["speech"] = resampler.forward(speech.squeeze(0)).numpy()
    batch["sampling_rate"] = resampler.new_freq
    return batch

def predict(data):
    features = processor(data["speech"], sampling_rate=data["sampling_rate"], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits
    
    decoded_results = []
    for logit in logits:
        pred_ids = torch.argmax(logit, dim=-1)
        mask = pred_ids.ge(1).unsqueeze(-1).expand(logit.size())
        vocab_size = logit.size()[-1]
        voice_prob = torch.nn.functional.softmax((torch.masked_select(logit, mask).view(-1,vocab_size)),dim=-1)
        gpt_input = torch.cat((torch.tensor([tokenizer.cls_token_id]).to(device),pred_ids[pred_ids>0]), 0)
        gpt_prob = torch.nn.functional.softmax(gpt_model(gpt_input).logits, dim=-1)[:voice_prob.size()[0],:]
        comb_pred_ids = torch.argmax(gpt_prob*voice_prob, dim=-1)
        decoded_results.append(processor.decode(comb_pred_ids))

    return decoded_results
```

Predict
```python
predict(load_file_to_data('voice file path'))
```

## Evaluation
The model can be evaluated as follows on the zh-tw test data of Common Voice.   
CER calculation refer to https://huggingface.co/ctl/wav2vec2-large-xlsr-cantonese 

env setup:
```
!pip install editdistance
!pip install torchaudio
!pip install datasets transformers
```

## Evaluation without LM:
```python
import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
)
import torch
import re
import sys
from transformers import AutoTokenizer, AutoModelWithLMHead 
from datasets import  Audio
from math import log

model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\"#$%&()*+,\-.\:;<=>?@\[\]\\\/^_`{|}~]"

tokenizer = AutoTokenizer.from_pretrained("ckiplab/gpt2-base-chinese")  
lm_model = AutoModelWithLMHead.from_pretrained("ckiplab/gpt2-base-chinese").to(device)
model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

ds = load_dataset("common_voice", 'zh-TW', split="test")
ds = ds.cast_column("audio", Audio(sampling_rate=16_000))
def map_to_array(batch):
    audio = batch["audio"]
    batch["speech"] = processor(audio["array"], sampling_rate=audio["sampling_rate"]).input_values[0]
    batch["sampling_rate"] = audio["sampling_rate"]
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'")
    return batch
ds = ds.map(map_to_array)

def map_to_pred(batch):
    features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits
    pred_ids = torch.argmax(logits, dim=-1)
    batch["predicted"] = processor.batch_decode(pred_ids)
    batch["target"] = batch["sentence"]
    return batch


result = ds.map(map_to_pred, batched=True, batch_size=3, remove_columns=list(ds.features.keys()))

def cer_cal(groundtruth, hypothesis):
    err = 0
    tot = 0
    for p, t in zip(hypothesis, groundtruth):
        err += float(ed.eval(p.lower(), t.lower()))
        tot += len(t)
    return err / tot
print("CER: {:2f}".format(100 * cer_cal(result["target"],result["predicted"])))
```

`CER: 28.70`.  
`TIME: 04:08 min`

## Evaluation with GPT:
```python
import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
)
import torch
import re
import sys
from transformers import AutoTokenizer, AutoModelWithLMHead 
from datasets import  Audio
from math import log

model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\"#$%&()*+,\-.\:;<=>?@\[\]\\\/^_`{|}~]"

tokenizer = AutoTokenizer.from_pretrained("ckiplab/gpt2-base-chinese")  
lm_model = AutoModelWithLMHead.from_pretrained("ckiplab/gpt2-base-chinese").to(device)
model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

ds = load_dataset("common_voice", 'zh-TW', split="test")
ds = ds.cast_column("audio", Audio(sampling_rate=16_000))
def map_to_array(batch):
    audio = batch["audio"]
    batch["speech"] = processor(audio["array"], sampling_rate=audio["sampling_rate"]).input_values[0]
    batch["sampling_rate"] = audio["sampling_rate"]
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'")
    return batch
ds = ds.map(map_to_array)

def map_to_pred(batch):
    features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits

    decoded_results = []
    for logit in logits:
        pred_ids = torch.argmax(logit, dim=-1)
        mask = pred_ids.ge(1).unsqueeze(-1).expand(logit.size())
        vocab_size = logit.size()[-1]
        voice_prob = torch.nn.functional.softmax((torch.masked_select(logit, mask).view(-1,vocab_size)),dim=-1)
        lm_input = torch.cat((torch.tensor([tokenizer.cls_token_id]).to(device),pred_ids[pred_ids>0]), 0)
        lm_prob = torch.nn.functional.softmax(lm_model(lm_input).logits, dim=-1)[:voice_prob.size()[0],:]
        comb_pred_ids = torch.argmax(lm_prob*voice_prob, dim=-1)
        decoded_results.append(processor.decode(comb_pred_ids))

    batch["predicted"] = decoded_results
    batch["target"] = batch["sentence"]
    return batch


result = ds.map(map_to_pred, batched=True, batch_size=3, remove_columns=list(ds.features.keys()))

def cer_cal(groundtruth, hypothesis):
    err = 0
    tot = 0
    for p, t in zip(hypothesis, groundtruth):
        err += float(ed.eval(p.lower(), t.lower()))
        tot += len(t)
    return err / tot
print("CER: {:2f}".format(100 * cer_cal(result["target"],result["predicted"])))
```

`CER 25.70`.  
`TIME: 06:04 min`


## Evaluation with GPT + beam search:
```python
import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
)
import torch
import re
import sys
from transformers import AutoTokenizer, AutoModelWithLMHead 
from datasets import  Audio
from math import log

model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\"#$%&()*+,\-.\:;<=>?@\[\]\\\/^_`{|}~]"

tokenizer = AutoTokenizer.from_pretrained("ckiplab/gpt2-base-chinese")  
lm_model = AutoModelWithLMHead.from_pretrained("ckiplab/gpt2-base-chinese").to(device)
model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

ds = load_dataset("common_voice", 'zh-TW', split="test")
ds = ds.cast_column("audio", Audio(sampling_rate=16_000))
def map_to_array(batch):
    audio = batch["audio"]
    batch["speech"] = processor(audio["array"], sampling_rate=audio["sampling_rate"]).input_values[0]
    batch["sampling_rate"] = audio["sampling_rate"]
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'")
    return batch
ds = ds.map(map_to_array)

def map_to_pred(batch):
    features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits
        
    decoded_results = []
    for logit in logits:
        sequences = [[[], 1.0]]
        pred_ids = torch.argmax(logit, dim=-1)
        mask = pred_ids.ge(1).unsqueeze(-1).expand(logit.size())
        vocab_size = logit.size()[-1]
        voice_prob = torch.nn.functional.softmax((torch.masked_select(logit, mask).view(-1,vocab_size)),dim=-1)
        while True:
            all_candidates = list()
            exceed = False
            for seq in sequences:
                tokens, score = seq
                gpt_input = torch.tensor([tokenizer.cls_token_id]+tokens).to(device)
                gpt_prob = torch.nn.functional.softmax(lm_model(gpt_input).logits, dim=-1)[:len(gpt_input),:]
                if len(gpt_input) >= len(voice_prob):
                    exceed = True
                comb_pred_ids = gpt_prob*voice_prob[:len(gpt_input)]
                v,i = torch.topk(comb_pred_ids,50,dim=-1)
                for tok_id,tok_prob in zip(i.tolist()[-1],v.tolist()[-1]):
                    candidate = [tokens + [tok_id], score + -log(tok_prob)]
                    all_candidates.append(candidate)
            ordered = sorted(all_candidates, key=lambda tup: tup[1])
            sequences = ordered[:10]
            if exceed:
                break
        decoded_results.append(processor.decode(sequences[0][0]))

    batch["predicted"] = decoded_results
    batch["target"] = batch["sentence"]
    return batch


result = ds.map(map_to_pred, batched=True, batch_size=3, remove_columns=list(ds.features.keys()))

def cer_cal(groundtruth, hypothesis):
    err = 0
    tot = 0
    for p, t in zip(hypothesis, groundtruth):
        err += float(ed.eval(p.lower(), t.lower()))
        tot += len(t)
    return err / tot
print("CER: {:2f}".format(100 * cer_cal(result["target"],result["predicted"])))
```

`CER 18.36`.  


## Evaluation with BERT:
```python
import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
)
import torch
import re
import sys
from transformers import AutoTokenizer, AutoModelForMaskedLM 

model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\"#$%&()*+,\-.\:;<=>?@\[\]\\\/^_`{|}~]"

tokenizer = AutoTokenizer.from_pretrained("bert-base-chinese")  
lm_model = AutoModelForMaskedLM.from_pretrained("bert-base-chinese").to(device)
model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

ds = load_dataset("common_voice", 'zh-TW', data_dir="./cv-corpus-6.1-2020-12-11", split="test")

resampler = torchaudio.transforms.Resample(orig_freq=48_000, new_freq=16_000)

def map_to_array(batch):
    speech, _ = torchaudio.load(batch["path"])
    batch["speech"] = resampler.forward(speech.squeeze(0)).numpy()
    batch["sampling_rate"] = resampler.new_freq
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'")
    return batch

ds = ds.map(map_to_array)

def map_to_pred(batch):
    features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits

    decoded_results = []
    for logit in logits:
        pred_ids = torch.argmax(logit, dim=-1)
        mask = ~pred_ids.eq(tokenizer.pad_token_id).unsqueeze(-1).expand(logit.size())
        vocab_size = logit.size()[-1]
        voice_prob = torch.nn.functional.softmax((torch.masked_select(logit, mask).view(-1,vocab_size)),dim=-1)
        lm_input = torch.masked_select(pred_ids, ~pred_ids.eq(tokenizer.pad_token_id)).unsqueeze(0)
        mask_lm_prob = voice_prob.clone()
        for i in range(lm_input.shape[-1]):
            masked_lm_input = lm_input.clone()
            masked_lm_input[0][i] = torch.tensor(tokenizer.mask_token_id).to('cuda')
            lm_prob = torch.nn.functional.softmax(lm_model(masked_lm_input).logits, dim=-1).squeeze(0)
            mask_lm_prob[i] = lm_prob[i]
        comb_pred_ids = torch.argmax(mask_lm_prob*voice_prob, dim=-1)
        decoded_results.append(processor.decode(comb_pred_ids))

    batch["predicted"] = decoded_results
    batch["target"] = batch["sentence"]
    return batch


result = ds.map(map_to_pred, batched=True, batch_size=1, remove_columns=list(ds.features.keys()))

def cer_cal(groundtruth, hypothesis):
    err = 0
    tot = 0
    for p, t in zip(hypothesis, groundtruth):
        err += float(ed.eval(p.lower(), t.lower()))
        tot += len(t)
    return err / tot
print("CER: {:2f}".format(100 * cer_cal(result["target"],result["predicted"])))
```
`CER 25.57`.  
`TIME: 09:49 min`

## Evaluation with T-TA:
setup
```
!git clone https://github.com/voidful/pytorch-tta.git
!mv ./pytorch-tta/tta ./tta
!wget https://github.com/voidful/pytorch-tta/releases/download/wiki_zh/wiki_zh.pt
```

```python
import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
)
import torch
import re
import sys
from tta.modeling_tta import TTALMModel
from transformers import AutoTokenizer
import torch



model_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
device = "cuda"
processor_name = "voidful/wav2vec2-large-xlsr-53-tw-gpt"
chars_to_ignore_regex = r"[¥•"#$%&'()*+,-/:;<=>@[\]^_`{|}~⦅⦆「」、 、〃〈〉《》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏﹑﹔·'℃°•·.﹑︰〈〉─《﹖﹣﹂﹁﹔!?。。"#$%&'()*+,﹐-/:;<=>@[\]^_`{|}~⦅⦆「」、、〃》「」『』【】〔〕〖〗〘〙〚〛〜〝〞〟〰〾〿–—‘’‛“”„‟…‧﹏..!\"#$%&()*+,\-.\:;<=>?@\[\]\\\/^_`{|}~]"

tokenizer = AutoTokenizer.from_pretrained("bert-base-chinese")  
lm_model = TTALMModel("bert-base-chinese")
tokenizer = AutoTokenizer.from_pretrained("bert-base-chinese")
lm_model.load_state_dict(torch.load("./wiki_zh.pt",map_location=torch.device('cuda')))
lm_model.to('cuda')
lm_model.eval()
model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

ds = load_dataset("common_voice", 'zh-TW', data_dir="./cv-corpus-6.1-2020-12-11", split="test")

resampler = torchaudio.transforms.Resample(orig_freq=48_000, new_freq=16_000)

def map_to_array(batch):
    speech, _ = torchaudio.load(batch["path"])
    batch["speech"] = resampler.forward(speech.squeeze(0)).numpy()
    batch["sampling_rate"] = resampler.new_freq
    batch["sentence"] = re.sub(chars_to_ignore_regex, '', batch["sentence"]).lower().replace("’", "'")
    return batch

ds = ds.map(map_to_array)

def map_to_pred(batch):
    features = processor(batch["speech"], sampling_rate=batch["sampling_rate"][0], padding=True, return_tensors="pt")
    input_values = features.input_values.to(device)
    attention_mask = features.attention_mask.to(device)
    with torch.no_grad():
        logits = model(input_values, attention_mask=attention_mask).logits

    decoded_results = []
    for logit in logits:
        pred_ids = torch.argmax(logit, dim=-1)
        mask = ~pred_ids.eq(tokenizer.pad_token_id).unsqueeze(-1).expand(logit.size())
        vocab_size = logit.size()[-1]
        voice_prob = torch.nn.functional.softmax((torch.masked_select(logit, mask).view(-1,vocab_size)),dim=-1)
        lm_input = torch.masked_select(pred_ids, ~pred_ids.eq(tokenizer.pad_token_id)).unsqueeze(0)
        lm_prob = torch.nn.functional.softmax(lm_model.forward(lm_input)[0], dim=-1).squeeze(0)
        comb_pred_ids = torch.argmax(lm_prob*voice_prob, dim=-1)
        decoded_results.append(processor.decode(comb_pred_ids))

    batch["predicted"] = decoded_results
    batch["target"] = batch["sentence"]
    return batch


result = ds.map(map_to_pred, batched=True, batch_size=16, remove_columns=list(ds.features.keys()))

def cer_cal(groundtruth, hypothesis):
    err = 0
    tot = 0
    for p, t in zip(hypothesis, groundtruth):
        err += float(ed.eval(p.lower(), t.lower()))
        tot += len(t)
    return err / tot
print("CER: {:2f}".format(100 * cer_cal(result["target"],result["predicted"])))
```

`CER: 25.77`.   
`TIME: 06:01 min`