From 0dad9f3b1cc7ab1600711d4a015b88ab65267184 Mon Sep 17 00:00:00 2001
From: Johannes Wagner <frankenjoe@hotmail.com>
Date: Wed, 6 Apr 2022 13:27:40 +0000
Subject: [PATCH] Update README.md

---
 README.md | 114 +++++++++++++++++++++++++++++++++++++++++++++++++++++-
 1 file changed, 113 insertions(+), 1 deletion(-)

diff --git a/README.md b/README.md
index 5f78d70..3ab0835 100644
--- a/README.md
+++ b/README.md
@@ -9,4 +9,116 @@ datasets:
 tags:
 - speech
 license: cc-by-nc-sa-4.0
----
\ No newline at end of file
+---
+
+# 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).
+
+# How to
+
+```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 = 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
+
+
+process_func(signal, sampling_rate)
+#  Arousal    dominance valence
+# [[0.5460759 0.6062269 0.4043165]]
+
+process_func(signal, sampling_rate, embeddings=True)
+# Pooled hidden states of last transformer layer
+# [[-0.00752167  0.0065819  -0.00746339 ...  0.00663631  0.00848747
+#   0.00599209]]
+```