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| language: en | |
| datasets: | |
| - msp-podcast | |
| inference: true | |
| tags: | |
| - speech | |
| - audio | |
| - wav2vec2 | |
| - audio-classification | |
| - emotion-recognition | |
| license: cc-by-nc-sa-4.0 | |
| pipeline_tag: audio-classification | |
| # Model for Dimensional Speech Emotion Recognition based on Wav2vec 2.0 | |
| The model expects a raw audio signal as input and outputs predictions for arousal, dominance and valence in a range of approximately 0...1. In addition, it also provides the pooled states of the last transformer layer. The model was created by fine-tuning [ | |
| Wav2Vec2-Large-Robust](https://huggingface.co/facebook/wav2vec2-large-robust) on [MSP-Podcast](https://ecs.utdallas.edu/research/researchlabs/msp-lab/MSP-Podcast.html) (v1.7). The model was pruned from 24 to 12 transformer layers before fine-tuning. An [ONNX](https://onnx.ai/") export of the model is available from [doi:10.5281/zenodo.6221127](https://zenodo.org/record/6221127). Further details are given in the associated [paper](https://arxiv.org/abs/2203.07378) and [tutorial](https://github.com/audeering/w2v2-how-to). | |
| # Usage | |
| ```python | |
| import numpy as np | |
| import torch | |
| import torch.nn as nn | |
| from transformers import Wav2Vec2Processor | |
| from transformers.models.wav2vec2.modeling_wav2vec2 import ( | |
| Wav2Vec2Model, | |
| Wav2Vec2PreTrainedModel, | |
| ) | |
| class RegressionHead(nn.Module): | |
| r"""Classification head.""" | |
| def __init__(self, config): | |
| super().__init__() | |
| self.dense = nn.Linear(config.hidden_size, config.hidden_size) | |
| self.dropout = nn.Dropout(config.final_dropout) | |
| self.out_proj = nn.Linear(config.hidden_size, config.num_labels) | |
| def forward(self, features, **kwargs): | |
| x = features | |
| x = self.dropout(x) | |
| x = self.dense(x) | |
| x = torch.tanh(x) | |
| x = self.dropout(x) | |
| x = self.out_proj(x) | |
| return x | |
| class EmotionModel(Wav2Vec2PreTrainedModel): | |
| r"""Speech emotion classifier.""" | |
| def __init__(self, config): | |
| super().__init__(config) | |
| self.config = config | |
| self.wav2vec2 = Wav2Vec2Model(config) | |
| self.classifier = RegressionHead(config) | |
| self.init_weights() | |
| def forward( | |
| self, | |
| input_values, | |
| ): | |
| outputs = self.wav2vec2(input_values) | |
| hidden_states = outputs[0] | |
| hidden_states = torch.mean(hidden_states, dim=1) | |
| logits = self.classifier(hidden_states) | |
| return hidden_states, logits | |
| # load model from hub | |
| device = 'cpu' | |
| model_name = 'audeering/wav2vec2-large-robust-12-ft-emotion-msp-dim' | |
| processor = Wav2Vec2Processor.from_pretrained(model_name) | |
| model = EmotionModel.from_pretrained(model_name) | |
| # dummy signal | |
| sampling_rate = 16000 | |
| signal = np.zeros((1, sampling_rate), dtype=np.float32) | |
| def process_func( | |
| x: np.ndarray, | |
| sampling_rate: int, | |
| embeddings: bool = False, | |
| ) -> np.ndarray: | |
| r"""Predict emotions or extract embeddings from raw audio signal.""" | |
| # run through processor to normalize signal | |
| # always returns a batch, so we just get the first entry | |
| # then we put it on the device | |
| y = processor(x, sampling_rate=sampling_rate) | |
| y = y['input_values'][0] | |
| y = y.reshape(1, -1) | |
| y = torch.from_numpy(y).to(device) | |
| # run through model | |
| with torch.no_grad(): | |
| y = model(y)[0 if embeddings else 1] | |
| # convert to numpy | |
| y = y.detach().cpu().numpy() | |
| return y | |
| print(process_func(signal, sampling_rate)) | |
| # Arousal dominance valence | |
| # [[0.5460754 0.6062266 0.40431657]] | |
| print(process_func(signal, sampling_rate, embeddings=True)) | |
| # Pooled hidden states of last transformer layer | |
| # [[-0.00752167 0.0065819 -0.00746342 ... 0.00663632 0.00848748 | |
| # 0.00599211]] | |
| ``` | |