#!/usr/bin/env python # coding: utf-8 # # Create embeddings with the transformer library # # We use the Huggingface transformers library to create an embedding for a an audio dataset # # # # ## tldr; Play as callable functions import datasets from transformers import AutoFeatureExtractor, AutoModel, ASTForAudioClassification import torch from renumics import spotlight import pandas as pd import umap import numpy as np _SPLIT = "train" def __set_device(): device = "cuda" if torch.cuda.is_available() else "cpu" if device == "cuda": torch.cuda.empty_cache() return device def extract_embeddings(model, feature_extractor): """Utility to compute embeddings.""" device = model.device def pp(batch): audios = [element["array"] for element in batch["audio"]] inputs = feature_extractor(raw_speech=audios, return_tensors="pt", padding=True).to(device) embeddings = model(**inputs).last_hidden_state[:, 0].cpu() return {"embedding": embeddings} return pp def huggingface_embedding(dataset, modelname, batched=True, batch_size=8): # initialize huggingface model feature_extractor = AutoFeatureExtractor.from_pretrained(modelname, padding=True) model = AutoModel.from_pretrained(modelname, output_hidden_states=True) #compute embedding device = __set_device() extract_fn = extract_embeddings(model.to(device), feature_extractor) updated_dataset = dataset.map(extract_fn, batched=batched, batch_size=batch_size) return updated_dataset def batch_probabilities(model, feature_extractor): device = model.device def processing(batch): audios = [element["array"] for element in batch["audio"]] inputs = feature_extractor(raw_speech=audios, return_tensors="pt", padding=True, sampling_rate=16000).to(device) outputs = model(**inputs) return {"logits": outputs.logits} return processing def annotate_probabilities(dataset, modelname, batched=True, batch_size= 8): model = ASTForAudioClassification.from_pretrained(modelname) feature_extractor = AutoFeatureExtractor.from_pretrained(modelname, padding=True) device = __set_device() calc_outputs = batch_probabilities (model.to(device), feature_extractor) output_dataset = dataset.map(calc_outputs, batched = batched, batch_size = batch_size) return output_dataset def annotate_batch(model, dataset): device = model.device def batch_annotation(batch): logits = [torch.tensor(element) for element in batch["logits"]] probabilities_per_class = [torch.nn.functional.softmax(logit, dim=-1) for logit in logits] predicted_class_ids = [torch.argmax(logit).item() for logit in logits] predicted_labels = [model.config.id2label[predicted_class_id] for predicted_class_id in predicted_class_ids] # pre-trained model to different amount of classes # -> id2label only reflects "internal label", not actual dataset label annotated_labels = [labels[element] for element in batch["label"]] probabilities = [] for index, prob_per_class in enumerate(probabilities_per_class): probabilities.append(prob_per_class[predicted_class_ids[index]].item()) return {"Probability": probabilities, "Predicted Label": predicted_labels, "Annotated Labels": annotated_labels, "Probability Vector": probabilities_per_class} return batch_annotation def annotate_dataset(dataset, modelname, batched=True, batch_size=8): model = ASTForAudioClassification.from_pretrained(modelname) device = __set_device() annotate = annotate_batch(model.to(device), dataset) annotated_dataset = dataset.map(annotate, batched=batched, batch_size=batch_size) return annotated_dataset # ## Step-by-step example on speech-commands # # ### Load speech-commands from Huggingface hub # Use validation split to evaluate model's performance on unseen data dataset = datasets.load_dataset('speech_commands', 'v0.01', split=_SPLIT) labels = dataset.features["label"].names # Let's have a look at all of the labels that we want to predict print(labels) # ### Compute probabilities and annotate dataset # First, calculate logits per sample # calculate logits for each sample and annotate dataset_annotated = annotate_probabilities(dataset, "MIT/ast-finetuned-speech-commands-v2") # Now annotate labels and probabilities dataset_annotated_complete = annotate_dataset(dataset_annotated, "MIT/ast-finetuned-speech-commands-v2") # ### Compute embedding with vision transformer from Huggingface dataset_enriched = huggingface_embedding(dataset_annotated_complete, "MIT/ast-finetuned-speech-commands-v2") # ### Reduce embeddings for faster visualization embeddings = np.stack(np.array(dataset_enriched['embedding'])) reducer = umap.UMAP() reduced_embedding = reducer.fit_transform(embeddings) dataset_enriched = dataset_enriched.add_column("embedding_reduced", list(reduced_embedding)) print(dataset_enriched.features) df = dataset_enriched.to_pandas() df.to_parquet("data/dataset_audio_" + _SPLIT + ".parquet.gzip", compression='gzip')