170 lines
4.6 KiB
Python
170 lines
4.6 KiB
Python
import torch
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import torch.nn as nn
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from torch.utils.data import Dataset
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from torch.utils.data import DataLoader
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from transformers import Wav2Vec2Processor
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from transformers.models.wav2vec2.modeling_wav2vec2 import (
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Wav2Vec2Model,
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Wav2Vec2PreTrainedModel,
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)
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import librosa
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import numpy as np
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import argparse
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from config import config
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import utils
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import os
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from tqdm import tqdm
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class RegressionHead(nn.Module):
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r"""Classification head."""
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def __init__(self, config):
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super().__init__()
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self.dense = nn.Linear(config.hidden_size, config.hidden_size)
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self.dropout = nn.Dropout(config.final_dropout)
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self.out_proj = nn.Linear(config.hidden_size, config.num_labels)
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def forward(self, features, **kwargs):
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x = features
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x = self.dropout(x)
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x = self.dense(x)
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x = torch.tanh(x)
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x = self.dropout(x)
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x = self.out_proj(x)
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return x
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class EmotionModel(Wav2Vec2PreTrainedModel):
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r"""Speech emotion classifier."""
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def __init__(self, config):
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super().__init__(config)
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self.config = config
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self.wav2vec2 = Wav2Vec2Model(config)
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self.classifier = RegressionHead(config)
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self.init_weights()
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def forward(
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self,
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input_values,
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):
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outputs = self.wav2vec2(input_values)
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hidden_states = outputs[0]
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hidden_states = torch.mean(hidden_states, dim=1)
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logits = self.classifier(hidden_states)
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return hidden_states, logits
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class AudioDataset(Dataset):
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def __init__(self, list_of_wav_files, sr, processor):
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self.list_of_wav_files = list_of_wav_files
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self.processor = processor
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self.sr = sr
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def __len__(self):
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return len(self.list_of_wav_files)
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def __getitem__(self, idx):
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wav_file = self.list_of_wav_files[idx]
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audio_data, _ = librosa.load(wav_file, sr=self.sr)
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processed_data = self.processor(audio_data, sampling_rate=self.sr)[
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"input_values"
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][0]
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return torch.from_numpy(processed_data)
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model_name = "./emotional/wav2vec2-large-robust-12-ft-emotion-msp-dim"
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processor = Wav2Vec2Processor.from_pretrained(model_name)
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model = EmotionModel.from_pretrained(model_name)
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def process_func(
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x: np.ndarray,
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sampling_rate: int,
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model: EmotionModel,
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processor: Wav2Vec2Processor,
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device: str,
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embeddings: bool = False,
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) -> np.ndarray:
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r"""Predict emotions or extract embeddings from raw audio signal."""
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model = model.to(device)
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y = processor(x, sampling_rate=sampling_rate)
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y = y["input_values"][0]
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y = torch.from_numpy(y).unsqueeze(0).to(device)
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# run through model
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with torch.no_grad():
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y = model(y)[0 if embeddings else 1]
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# convert to numpy
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y = y.detach().cpu().numpy()
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return y
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def get_emo(path):
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wav, sr = librosa.load(path, 16000)
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device = config.bert_gen_config.device
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return process_func(
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np.expand_dims(wav, 0).astype(np.float),
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sr,
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model,
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processor,
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device,
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embeddings=True,
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).squeeze(0)
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if __name__ == "__main__":
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parser = argparse.ArgumentParser()
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parser.add_argument(
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"-c", "--config", type=str, default=config.bert_gen_config.config_path
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)
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parser.add_argument(
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"--num_processes", type=int, default=config.bert_gen_config.num_processes
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)
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args, _ = parser.parse_known_args()
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config_path = args.config
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hps = utils.get_hparams_from_file(config_path)
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device = config.bert_gen_config.device
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model_name = "./emotional/wav2vec2-large-robust-12-ft-emotion-msp-dim"
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processor = (
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Wav2Vec2Processor.from_pretrained(model_name)
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if processor is None
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else processor
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)
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model = (
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EmotionModel.from_pretrained(model_name).to(device)
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if model is None
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else model.to(device)
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)
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lines = []
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with open(hps.data.training_files, encoding="utf-8") as f:
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lines.extend(f.readlines())
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with open(hps.data.validation_files, encoding="utf-8") as f:
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lines.extend(f.readlines())
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wavnames = [line.split("|")[0] for line in lines]
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dataset = AudioDataset(wavnames, 16000, processor)
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data_loader = DataLoader(dataset, batch_size=1, shuffle=False, num_workers=16)
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with torch.no_grad():
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for i, data in tqdm(enumerate(data_loader), total=len(data_loader)):
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wavname = wavnames[i]
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emo_path = wavname.replace(".wav", ".emo.npy")
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if os.path.exists(emo_path):
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continue
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emb = model(data.to(device))[0].detach().cpu().numpy()
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np.save(emo_path, emb)
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print("Emo vec 生成完毕!")
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