--- language: - ko # Example: fr license: apache-2.0 # Example: apache-2.0 or any license from https://hf.co/docs/hub/repositories-licenses library_name: transformers # Optional. Example: keras or any library from https://github.com/huggingface/hub-docs/blob/main/js/src/lib/interfaces/Libraries.ts tags: - audio - automatic-speech-recognition datasets: - KsponSpeech metrics: - wer # Example: wer. Use metric id from https://hf.co/metrics --- # ko-spelling-wav2vec2-conformer-del-1s ## Table of Contents - [ko-spelling-wav2vec2-conformer-del-1s](#ko-spelling-wav2vec2-conformer-del-1s) - [Table of Contents](#table-of-contents) - [Model Details](#model-details) - [Evaluation](#evaluation) - [How to Get Started With the Model](#how-to-get-started-with-the-model) ## Model Details - **Model Description:** 해당 모델은 wav2vec2-conformer base architecture에 scratch pre-training 되었습니다.
Wav2Vec2ConformerForCTC를 이용하여 KsponSpeech에 대한 Fine-Tuning 모델입니다.
- Dataset use [AIHub KsponSpeech](https://www.aihub.or.kr/aihubdata/data/view.do?currMenu=115&topMenu=100&aihubDataSe=realm&dataSetSn=123)
Datasets는 해당 Data를 전처리하여 임의로 만들어 사용하였습니다.
해당 모델은 **철자전사** 기준의 데이터로 학습된 모델입니다. (숫자와 영어는 각 표기법을 따름)
- **Developed by:** TADev (@lIlBrother, @ddobokki, @jp42maru) - **Language(s):** Korean - **License:** apache-2.0 - **Parent Model:** See the [wav2vec2-conformer](https://huggingface.co/docs/transformers/model_doc/wav2vec2-conformer) for more information about the pre-trained base model. (해당 모델은 wav2vec2-conformer base architecture에 scratch pre-training 되었습니다.) ## Evaluation Just using `load_metric("wer")` and `load_metric("wer")` in huggingface `datasets` library
## How to Get Started With the Model ```python from transformers import ( AutoConfig, AutoFeatureExtractor, AutoModelForCTC, AutoTokenizer, Wav2Vec2ProcessorWithLM, ) from transformers.pipelines import AutomaticSpeechRecognitionPipeline import librosa # 모델과 토크나이저, 예측을 위한 각 모듈들을 불러옵니다. config = AutoConfig.from_pretrained(model_config_path) model = AutoModelForCTC.from_pretrained( model_name_or_path, config=config, ) feature_extractor = AutoFeatureExtractor.from_pretrained(model_name_or_path) tokenizer = AutoTokenizer.from_pretrained(model_name_or_path) beamsearch_decoder = build_ctcdecoder( labels=list(tokenizer.encoder.keys()), kenlm_model_path=None, ) processor = Wav2Vec2ProcessorWithLM( feature_extractor=feature_extractor, tokenizer=tokenizer, decoder=beamsearch_decoder ) # 실제 예측을 위한 파이프라인에 정의된 모듈들을 삽입. asr_pipeline = AutomaticSpeechRecognitionPipeline( model=model, tokenizer=processor.tokenizer, feature_extractor=processor.feature_extractor, decoder=processor.decoder, device=-1, ) # 음성파일을 불러오고 beamsearch 파라미터를 특정하여 예측을 수행합니다. raw_data, _ = librosa.load(audio_path, sr=16000) kwargs = {"decoder_kwargs": {"beam_width": 100}} pred = asr_pipeline(inputs=raw_data, **kwargs)["text"] # 모델이 자소 분리 유니코드 텍스트로 나오므로, 일반 String으로 변환해줄 필요가 있습니다. result = unicodedata.normalize("NFC", pred) print(result) # 안녕하세요 123 테스트입니다. ``` *Beam-100 Result (WER)*: | "clean" | "other" | | ------- | ------- | | 22.01 | 27.34 |