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@@ -11,170 +11,42 @@ tags:
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  - speech
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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 Spanish by Jonatas Grosman
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- results:
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- - task:
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- name: Speech Recognition
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- type: automatic-speech-recognition
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- dataset:
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- name: Common Voice es
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- type: common_voice
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- args: es
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- metrics:
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- - name: Test WER
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- type: wer
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- value: 8.81
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- - name: Test CER
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- type: cer
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- value: 2.70
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  ---
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- # Wav2Vec2-Large-XLSR-53-Spanish
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- Fine-tuned [facebook/wav2vec2-large-xlsr-53](https://huggingface.co/facebook/wav2vec2-large-xlsr-53) on Spanish using the [Common Voice](https://huggingface.co/datasets/common_voice).
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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 thanks to the GPU credits generously given by the [OVHcloud](https://www.ovhcloud.com/en/public-cloud/ai-training/) :)
 
 
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- The script used for training can be found here: https://github.com/jonatasgrosman/wav2vec2-sprint
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- ## Usage
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- The model can be used directly (without a language model) as follows...
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-
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- Using the [ASRecognition](https://github.com/jonatasgrosman/asrecognition) library:
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-
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- ```python
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- from asrecognition import ASREngine
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-
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- asr = ASREngine("es", model_path="jonatasgrosman/wav2vec2-large-xlsr-53-spanish")
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-
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- audio_paths = ["/path/to/file.mp3", "/path/to/another_file.wav"]
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- transcriptions = asr.transcribe(audio_paths)
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- ```
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-
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- Writing your own inference script:
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-
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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 = "es"
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- MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-spanish"
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- SAMPLES = 10
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-
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- test_dataset = load_dataset("common_voice", LANG_ID, split=f"test[:{SAMPLES}]")
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-
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- processor = Wav2Vec2Processor.from_pretrained(MODEL_ID)
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- model = Wav2Vec2ForCTC.from_pretrained(MODEL_ID)
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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 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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-
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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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-
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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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-
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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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-
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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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-
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- | Reference | Prediction |
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- | ------------- | ------------- |
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- | HABITA EN AGUAS POCO PROFUNDAS Y ROCOSAS. | HABITAN AGUAS POCO PROFUNDAS Y ROCOSAS |
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- | OPERA PRINCIPALMENTE VUELOS DE CABOTAJE Y REGIONALES DE CARGA. | OPERA PRINCIPALMENTE VUELO DE CARBOTAJES Y REGIONALES DE CARGAN |
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- | PARA VISITAR CONTACTAR PRIMERO CON LA DIRECCIÓN. | PARA VISITAR CONTACTAR PRIMERO CON LA DIRECCIÓN |
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- | TRES | TRES |
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- | REALIZÓ LOS ESTUDIOS PRIMARIOS EN FRANCIA, PARA CONTINUAR LUEGO EN ESPAÑA. | REALIZÓ LOS ESTUDIOS PRIMARIOS EN FRANCIA PARA CONTINUAR LUEGO EN ESPAÑA |
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- | EN LOS AÑOS QUE SIGUIERON, ESTE TRABAJO ESPARTA PRODUJO DOCENAS DE BUENOS JUGADORES. | EN LOS AÑOS QUE SIGUIERON ESTE TRABAJO ESPARTA PRODUJO DOCENA DE BUENOS JUGADORES |
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- | SE ESTÁ TRATANDO DE RECUPERAR SU CULTIVO EN LAS ISLAS CANARIAS. | SE ESTÓ TRATANDO DE RECUPERAR SU CULTIVO EN LAS ISLAS CANARIAS |
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- | SÍ | SÍ |
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- | "FUE ""SACADA"" DE LA SERIE EN EL EPISODIO ""LEAD"", EN QUE ALEXANDRA CABOT REGRESÓ." | FUE SACADA DE LA SERIE EN EL EPISODIO LEED EN QUE ALEXANDRA KAOT REGRESÓ |
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- | SE UBICAN ESPECÍFICAMENTE EN EL VALLE DE MOKA, EN LA PROVINCIA DE BIOKO SUR. | SE UBICAN ESPECÍFICAMENTE EN EL VALLE DE MOCA EN LA PROVINCIA DE PÍOCOSUR |
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-
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- ## Evaluation
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-
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- The model can be evaluated as follows on the Spanish test data of Common Voice.
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-
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- ```python
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- import torch
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- import re
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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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-
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- LANG_ID = "es"
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- MODEL_ID = "jonatasgrosman/wav2vec2-large-xlsr-53-spanish"
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- DEVICE = "cuda"
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-
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- CHARS_TO_IGNORE = [",", "?", "¿", ".", "!", "¡", ";", ";", ":", '""', "%", '"', "�", "ʿ", "·", "჻", "~", "՞",
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- "؟", "،", "।", "॥", "«", "»", "„", "“", "”", "「", "」", "‘", "’", "《", "》", "(", ")", "[", "]",
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- "{", "}", "=", "`", "_", "+", "<", ">", "…", "–", "°", "´", "ʾ", "‹", "›", "©", "®", "—", "→", "。",
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- "、", "﹂", "﹁", "‧", "~", "﹏", ",", "{", "}", "(", ")", "[", "]", "【", "】", "‥", "〽",
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- "『", "』", "〝", "〟", "⟨", "⟩", "〜", ":", "!", "?", "♪", "؛", "/", "\\", "º", "−", "^", "ʻ", "ˆ"]
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-
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- test_dataset = load_dataset("common_voice", LANG_ID, split="test")
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-
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- wer = load_metric("wer.py") # https://github.com/jonatasgrosman/wav2vec2-sprint/blob/main/wer.py
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- cer = load_metric("cer.py") # https://github.com/jonatasgrosman/wav2vec2-sprint/blob/main/cer.py
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-
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- chars_to_ignore_regex = f"[{re.escape(''.join(CHARS_TO_IGNORE))}]"
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-
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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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-
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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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- with warnings.catch_warnings():
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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"] = re.sub(chars_to_ignore_regex, "", batch["sentence"]).upper()
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- return batch
151
-
152
- test_dataset = test_dataset.map(speech_file_to_array_fn)
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154
- # Preprocessing the datasets.
155
- # We need to read the audio files as arrays
156
- def evaluate(batch):
157
- inputs = processor(batch["speech"], sampling_rate=16_000, return_tensors="pt", padding=True)
158
 
159
- with torch.no_grad():
160
- logits = model(inputs.input_values.to(DEVICE), attention_mask=inputs.attention_mask.to(DEVICE)).logits
161
 
162
- 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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166
- result = test_dataset.map(evaluate, batched=True, batch_size=8)
 
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168
- predictions = [x.upper() for x in result["pred_strings"]]
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- references = [x.upper() for x in result["sentence"]]
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- print(f"WER: {wer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}")
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- print(f"CER: {cer.compute(predictions=predictions, references=references, chunk_size=1000) * 100}")
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  ```
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- **Test Result**:
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-
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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 2021-04-22). 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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  | ------------- | ------------- | ------------- |
 
11
  - speech
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  - xlsr-fine-tuning-week
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  license: apache-2.0
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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  ---
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+ # Wav2Vec2-Large-XLSR-53-Spanish-With-LM
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+ This is a model copy of [Wav2Vec2-Large-XLSR-53-Spanish](https://huggingface.co/jonatasgrosman/wav2vec2-large-xlsr-53-spanish)
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+ that has language model support.
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+ This model card can be seen as a demo for the [pyctcdecode](https://github.com/kensho-technologies/pyctcdecode) integration
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+ with Transformers led by [this PR](https://github.com/huggingface/transformers/pull/14339). The PR explains in-detail how the
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+ integration works.
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+ In a nutshell: This PR adds a new Wav2Vec2WithLMProcessor class as drop-in replacement for Wav2Vec2Processor.
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+ The only change from the existing ASR pipeline will be:
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+ ```diff
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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  from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
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+ from datasets import load_dataset
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+ ds = load_dataset("common_voice", "es", split="test", streaming=True)
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
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+ sample = next(iter(ds))
 
 
 
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+ model = Wav2Vec2ForCTC.from_pretrained("patrickvonplaten/wav2vec2-large-xlsr-53-spanish-with-lm")
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+ processor = Wav2Vec2Processor.from_pretrained("patrickvonplaten/wav2vec2-large-xlsr-53-spanish-with-lm")
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+ input_values = processor(sample["audio"]["array"], return_tensors="pt").input_values
 
 
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+ logits = model(input_values).logits
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+ prediction_ids = torch.argmax(logits, dim=-1)
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+ transcription = processor.batch_decode(prediction_ids)
 
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+ print(transcription)
 
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  ```
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  | Model | WER | CER |
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  | ------------- | ------------- | ------------- |