Wav2vec2 Xlsr Multilingual 56

環境設定：

!pip install torchaudio
!pip install datasets transformers
!pip install asrp
!wget -O lang_ids.pk https://huggingface.co/voidful/wav2vec2-xlsr-multilingual-56/raw/main/lang_ids.pk

使用方法

import torchaudio
from datasets import load_dataset, load_metric
from transformers import (
    Wav2Vec2ForCTC,
    Wav2Vec2Processor,
    AutoTokenizer, 
    AutoModelWithLMHead 
)
import torch
import re
import sys
import soundfile as sf
model_name = "voidful/wav2vec2-xlsr-multilingual-56"
device = "cuda"
processor_name = "voidful/wav2vec2-xlsr-multilingual-56"

import pickle
with open("lang_ids.pk", 'rb') as output:
    lang_ids = pickle.load(output)
    
model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device)
processor = Wav2Vec2Processor.from_pretrained(processor_name)

model.eval()

def load_file_to_data(file,sampling_rate=16_000):
    batch = {}
    speech, _ = torchaudio.load(file)
    if sampling_rate != '16_000' or sampling_rate != '16000':
        resampler = torchaudio.transforms.Resample(orig_freq=sampling_rate, new_freq=16_000)
        batch["speech"] = resampler.forward(speech.squeeze(0)).numpy()
        batch["sampling_rate"] = resampler.new_freq
    else:
        batch["speech"] = speech.squeeze(0).numpy()
        batch["sampling_rate"] = '16000'
    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)
            comb_pred_ids = torch.argmax(voice_prob, dim=-1)
            decoded_results.append(processor.decode(comb_pred_ids))

    return decoded_results

def predict_lang_specific(data,lang_code):
    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.eq(processor.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)
            filtered_input = pred_ids[pred_ids!=processor.tokenizer.pad_token_id].view(1,-1).to(device)
            if len(filtered_input[0]) == 0:
                decoded_results.append("")
            else:
                lang_mask = torch.empty(voice_prob.shape[-1]).fill_(0)
                lang_index = torch.tensor(sorted(lang_ids[lang_code]))
                lang_mask.index_fill_(0, lang_index, 1)
                lang_mask = lang_mask.to(device)
                comb_pred_ids = torch.argmax(lang_mask*voice_prob, dim=-1)
                decoded_results.append(processor.decode(comb_pred_ids))
                
    return decoded_results


predict(load_file_to_data('audio file path',sampling_rate=16_000)) # オーディオファイルのサンプリングレートに注意

predict_lang_specific(load_file_to_data('audio file path',sampling_rate=16_000),'en') # オーディオファイルのサンプリングレートに注意