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README.md
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- xlsr-fine-tuning-week
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license: apache-2.0
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model-index:
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- name: XLSR Wav2Vec2 Chinese (zh-CN) by
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results:
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- task:
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name: Speech Recognition
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metrics:
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- name: Test WER
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type: wer
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value:
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- name: Test CER
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type: cer
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value:
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---
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# Fine-tuned XLSR-53 large model for speech recognition in Chinese
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Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Chinese using the train and validation splits of [Common Voice 6.1](https://huggingface.co/datasets/common_voice), [CSS10](https://github.com/Kyubyong/css10) and [ST-CMDS](http://www.openslr.org/38/).
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When using this model, make sure that your speech input is sampled at 16kHz.
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This model has been fine-tuned
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The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint
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```python
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from huggingsound import SpeechRecognitionModel
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model = SpeechRecognitionModel("
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audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
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transcriptions = model.transcribe(audio_paths)
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```
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Writing your own inference script:
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```python
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import torch
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import librosa
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from datasets import load_dataset
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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LANG_ID = "zh-CN"
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MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-chinese-zh-cn"
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SAMPLES = 10
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test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
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processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
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model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
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# Preprocessing the datasets.
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# We need to read the audio files as arrays
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def speech_file_to_array_fn(batch):
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speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
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batch["speech"] = speech_array
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batch["sentence"] = batch["sentence"].upper()
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return batch
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test_dataset = test_dataset.map(speech_file_to_array_fn)
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inputs = processor(test_dataset["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
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with torch.no_grad():
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logits = model(inputs.input_values, attention_mask=inputs.attention_mask).logits
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predicted_ids = torch.argmax(logits, dim=-1)
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predicted_sentences = processor.batch_decode(predicted_ids)
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for i, predicted_sentence in enumerate(predicted_sentences):
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print("-" * 100)
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print("Reference:", test_dataset[i]["sentence"])
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print("Prediction:", predicted_sentence)
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```
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| Reference | Prediction |
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| ------------- | ------------- |
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| 宋朝末年年间定居粉岭围。 | 宋朝末年年间定居分定为 |
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| 渐渐行动不便 | 建境行动不片 |
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| 二十一年去世。 | 二十一年去世 |
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| 他们自称恰哈拉。 | 他们自称家哈<unk> |
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| 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<unk> |
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| 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 |
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| 这一名称一直沿用至今。 | 这一名称一直沿用是心 |
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| 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 |
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| 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 |
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| 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 |
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## Evaluation
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import librosa
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from datasets import load_dataset, load_metric
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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LANG_ID = "zh-CN"
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MODEL_ID = "
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DEVICE = "cuda"
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"؟", "،", "।", "॥", "«", "»", "„", "“", "”", "「", "」", "‘", "’", "《", "》", "(", ")", "[", "]",
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"{", "}", "=", "`", "_", "+", "<", ">", "…", "–", "°", "´", "ʾ", "‹", "›", "©", "®", "—", "→", "。",
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"、", "﹂", "﹁", "‧", "~", "﹏", ",", "{", "}", "(", ")", "[", "]", "【", "】", "‥", "〽",
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"『", "』", "〝", "〟", "⟨", "⟩", "〜", ":", "!", "?", "♪", "؛", "/", "\\", "º", "−", "^", "'", "ʻ", "ˆ"]
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test_dataset = load_dataset("common_voice", LANG_ID, split="test")
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processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
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model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
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model.to(DEVICE)
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# Preprocessing the datasets.
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# We need to read the audio files as arrays
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def speech_file_to_array_fn(batch):
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warnings.simplefilter("ignore")
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speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
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batch["speech"] = speech_array
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batch["sentence"] =
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return batch
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# Preprocessing the datasets.
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# We need to read the audio files as arrays
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def evaluate(batch):
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inputs = processor(
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with torch.no_grad():
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logits = model(
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pred_ids = torch.argmax(logits, dim=-1)
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batch["pred_strings"] = processor.batch_decode(pred_ids)
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return batch
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result = test_dataset.map(evaluate, batched=True, batch_size=8)
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```
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**Test Result**:
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In the table below I report the Word Error Rate (WER) and the Character Error Rate (CER) of the model. I ran the evaluation script described above on other models as well (on
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| Model | WER | CER |
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| ------------- | ------------- | ------------- |
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| jonatasgrosman/wav2vec2-large-xlsr-53-chinese-zh-cn | **82.37%** | **19.03%** |
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| ydshieh/wav2vec2-large-xlsr-53-chinese-zh-cn-gpt | 84.01% | 20.95% |
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@misc{grosman2021xlsr53-large-chinese,
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title={Fine-tuned {XLSR}-53 large model for speech recognition in {C}hinese},
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author={Grosman, Jonatas},
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howpublished={\url{https://huggingface.co/
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year={2021}
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}
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```
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- xlsr-fine-tuning-week
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license: apache-2.0
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model-index:
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- name: XLSR Wav2Vec2 Chinese (zh-CN) by wbbbbb
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results:
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- task:
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name: Speech Recognition
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metrics:
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- name: Test WER
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type: wer
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value: 70.47
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- name: Test CER
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type: cer
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value: 12.30
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---
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# Fine-tuned XLSR-53 large model for speech recognition in Chinese
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Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Chinese using the train and validation splits of [Common Voice 6.1](https://huggingface.co/datasets/common_voice), [CSS10](https://github.com/Kyubyong/css10) and [ST-CMDS](http://www.openslr.org/38/).
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When using this model, make sure that your speech input is sampled at 16kHz.
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This model has been fine-tuned on RTX3090 for 50h
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The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint
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```python
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from huggingsound import SpeechRecognitionModel
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model = SpeechRecognitionModel("wbbbbb/wav2vec2-large-chinese-zh-cn")
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audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
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transcriptions = model.transcribe(audio_paths)
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```
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## Evaluation
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import librosa
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from datasets import load_dataset, load_metric
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from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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import warnings
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import os
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os.environ["KMP_AFFINITY"] = ""
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LANG_ID = "zh-CN"
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MODEL_ID = "zh-CN-output-aishell"
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DEVICE = "cuda"
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test_dataset = load_dataset("common_voice", LANG_ID, split="test")
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wer = load_metric("wer")
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cer = load_metric("cer")
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processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
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model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
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model.to(DEVICE)
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# Preprocessing the datasets.
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# We need to read the audio files as arrays
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def speech_file_to_array_fn(batch):
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warnings.simplefilter("ignore")
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speech_array, sampling_rate = librosa.load(batch["path"], sr=16_000)
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batch["speech"] = speech_array
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batch["sentence"] = (
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re.sub("([^\u4e00-\u9fa5\u0030-\u0039])", "", batch["sentence"]).lower() + " "
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)
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return batch
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test_dataset = test_dataset.map(
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speech_file_to_array_fn,
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num_proc=15,
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remove_columns=['client_id', 'up_votes', 'down_votes', 'age', 'gender', 'accent', 'locale', 'segment'],
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)
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# Preprocessing the datasets.
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# We need to read the audio files as arrays
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def evaluate(batch):
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inputs = processor(
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batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True
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)
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with torch.no_grad():
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logits = model(
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inputs.input_values.to(DEVICE),
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attention_mask=inputs.attention_mask.to(DEVICE),
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).logits
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pred_ids = torch.argmax(logits, dim=-1)
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batch["pred_strings"] = processor.batch_decode(pred_ids)
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return batch
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result = test_dataset.map(evaluate, batched=True, batch_size=8)
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predictions = [x.lower() for x in result["pred_strings"]]
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references = [x.lower() for x in result["sentence"]]
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print(
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f"WER: {wer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}"
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)
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print(f"CER: {cer.compute(predictions=predictions, references=references) * 100}")
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```
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**Test Result**:
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In the table below I report the Word Error Rate (WER) and the Character Error Rate (CER) of the model. I ran the evaluation script described above on other models as well (on 2022-07-18). Note that the table below may show different results from those already reported, this may have been caused due to some specificity of the other evaluation scripts used.
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| Model | WER | CER |
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| ------------- | ------------- | ------------- |
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| wbbbbb/wav2vec2-large-chinese-zh-cn | **70.47%** | **12.30%** |
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| jonatasgrosman/wav2vec2-large-xlsr-53-chinese-zh-cn | **82.37%** | **19.03%** |
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| ydshieh/wav2vec2-large-xlsr-53-chinese-zh-cn-gpt | 84.01% | 20.95% |
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@misc{grosman2021xlsr53-large-chinese,
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title={Fine-tuned {XLSR}-53 large model for speech recognition in {C}hinese},
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author={Grosman, Jonatas},
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howpublished={\url{https://huggingface.co/wbbbbb/wav2vec2-large-chinese-zh-cn}},
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year={2021}
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}
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```
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