import gradio as gr import nltk import librosa from transformers import pipeline from transformers.file_utils import cached_path, hf_bucket_url import os, zipfile from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC, Wav2Vec2Tokenizer from datasets import load_dataset import torch import kenlm import torchaudio from pyctcdecode import Alphabet, BeamSearchDecoderCTC, LanguageModel """Vietnamese speech2text""" cache_dir = './cache/' processor = Wav2Vec2Processor.from_pretrained("nguyenvulebinh/wav2vec2-base-vietnamese-250h", cache_dir=cache_dir) vi_model = Wav2Vec2ForCTC.from_pretrained("nguyenvulebinh/wav2vec2-base-vietnamese-250h", cache_dir=cache_dir) lm_file = hf_bucket_url("nguyenvulebinh/wav2vec2-base-vietnamese-250h", filename='vi_lm_4grams.bin.zip') lm_file = cached_path(lm_file,cache_dir=cache_dir) with zipfile.ZipFile(lm_file, 'r') as zip_ref: zip_ref.extractall(cache_dir) lm_file = cache_dir + 'vi_lm_4grams.bin' def get_decoder_ngram_model(tokenizer, ngram_lm_path): vocab_dict = tokenizer.get_vocab() sort_vocab = sorted((value, key) for (key, value) in vocab_dict.items()) vocab = [x[1] for x in sort_vocab][:-2] vocab_list = vocab # convert ctc blank character representation vocab_list[tokenizer.pad_token_id] = "" # replace special characters vocab_list[tokenizer.unk_token_id] = "" # vocab_list[tokenizer.bos_token_id] = "" # vocab_list[tokenizer.eos_token_id] = "" # convert space character representation vocab_list[tokenizer.word_delimiter_token_id] = " " # specify ctc blank char index, since conventially it is the last entry of the logit matrix alphabet = Alphabet.build_alphabet(vocab_list, ctc_token_idx=tokenizer.pad_token_id) lm_model = kenlm.Model(ngram_lm_path) decoder = BeamSearchDecoderCTC(alphabet, language_model=LanguageModel(lm_model)) return decoder ngram_lm_model = get_decoder_ngram_model(processor.tokenizer, lm_file) # define function to read in sound file def speech_file_to_array_fn(path, max_seconds=10): batch = {"file": path} speech_array, sampling_rate = torchaudio.load(batch["file"]) if sampling_rate != 16000: transform = torchaudio.transforms.Resample(orig_freq=sampling_rate, new_freq=16000) speech_array = transform(speech_array) speech_array = speech_array[0] if max_seconds > 0: speech_array = speech_array[:max_seconds*16000] batch["speech"] = speech_array.numpy() batch["sampling_rate"] = 16000 return batch # tokenize def speech2text_vi(audio): # read in sound file # load dummy dataset and read soundfiles ds = speech_file_to_array_fn(audio.name) # infer model input_values = processor( ds["speech"], sampling_rate=ds["sampling_rate"], return_tensors="pt" ).input_values # decode ctc output logits = vi_model(input_values).logits[0] pred_ids = torch.argmax(logits, dim=-1) greedy_search_output = processor.decode(pred_ids) beam_search_output = ngram_lm_model.decode(logits.cpu().detach().numpy(), beam_width=500) return beam_search_output """Machine translation""" vien_model_checkpoint = "datnth1709/finetuned_HelsinkiNLP-opus-mt-vi-en_PhoMT" vien_translator = pipeline("translation", model=vien_model_checkpoint) def translate_vi2en(Vietnamese): return vien_translator(Vietnamese)[0]['translation_text'] """ Inference""" def inference_vien(audio): vi_text = speech2text_vi(audio) en_text = translate_vi2en(vi_text) return vi_text, en_text def transcribe_vi_1(audio, state_en=""): ds = speech_file_to_array_fn(audio.name) # infer model input_values = processor( ds["speech"], sampling_rate=ds["sampling_rate"], return_tensors="pt" ).input_values # decode ctc output logits = vi_model(input_values).logits[0] pred_ids = torch.argmax(logits, dim=-1) greedy_search_output = processor.decode(pred_ids) beam_search_output = ngram_lm_model.decode(logits.cpu().detach().numpy(), beam_width=500) en_text = translate_vi2en(beam_search_output) state_en += en_text + " " return state_en, state_en """Gradio demo""" vi_example_text = ["Có phải bạn đang muốn tìm mua nhà ở ngoại ô thành phố Hồ Chí Minh không?", "Ánh mắt ta chạm nhau. Chỉ muốn ngắm anh lâu thật lâu.", "Nếu như một câu nói có thể khiến em vui."] vi_example_voice =[['vi_speech_01.wav'], ['vi_speech_02.wav'], ['vi_speech_03.wav']] gr.Interface( fn=transcribe_vi_1, inputs=[ gr.Audio(source="microphone", label="Input Vietnamese Audio", type="file", streaming=True), "state", ], outputs= [ "text", "state", ], live=True).launch()