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  1. README.md +93 -0
  2. config.json +91 -0
  3. preprocessor_config.json +9 -0
  4. pytorch_model.bin +3 -0
README.md ADDED
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+ ---
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+ language: en
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+ datasets:
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+ - superb
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+ tags:
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+ - speech
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+ - audio
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+ - hubert
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+ - audio-classification
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+ license: apache-2.0
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+ ---
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+
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+ # Hubert-Large for Emotion Recognition
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+
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+ ## Model description
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+
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+ This is a ported version of
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+ [S3PRL's Hubert for the SUPERB Emotion Recognition task](https://github.com/s3prl/s3prl/tree/master/s3prl/downstream/emotion).
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+
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+ The base model is [hubert-large-ll60k](https://huggingface.co/facebook/hubert-large-ll60k), which is pretrained on 16kHz
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+ sampled speech audio. When using the model make sure that your speech input is also sampled at 16Khz.
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+
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+ For more information refer to [SUPERB: Speech processing Universal PERformance Benchmark](https://arxiv.org/abs/2105.01051)
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+
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+ ## Task and dataset description
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+
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+ Emotion Recognition (ER) predicts an emotion class for each utterance. The most widely used ER dataset
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+ [IEMOCAP](https://sail.usc.edu/iemocap/) is adopted, and we follow the conventional evaluation protocol:
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+ we drop the unbalanced emotion classes to leave the final four classes with a similar amount of data points and
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+ cross-validate on five folds of the standard splits.
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+
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+ For the original model's training and evaluation instructions refer to the
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+ [S3PRL downstream task README](https://github.com/s3prl/s3prl/tree/master/s3prl/downstream#er-emotion-recognition).
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+
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+
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+ ## Usage examples
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+
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+ You can use the model via the Audio Classification pipeline:
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+ ```python
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+ from datasets import load_dataset
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+ from transformers import pipeline
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+
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+ dataset = load_dataset("anton-l/superb_demo", "er", split="session1")
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+
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+ classifier = pipeline("audio-classification", model="superb/hubert-large-superb-er")
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+ labels = classifier(dataset[0]["file"], top_k=5)
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+ ```
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+
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+ Or use the model directly:
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+ ```python
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+ import torch
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+ import librosa
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+ from datasets import load_dataset
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+ from transformers import HubertForSequenceClassification, Wav2Vec2FeatureExtractor
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+
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+ def map_to_array(example):
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+ speech, _ = librosa.load(example["file"], sr=16000, mono=True)
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+ example["speech"] = speech
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+ return example
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+
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+ # load a demo dataset and read audio files
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+ dataset = load_dataset("anton-l/superb_demo", "er", split="session1")
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+ dataset = dataset.map(map_to_array)
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+
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+ model = HubertForSequenceClassification.from_pretrained("superb/hubert-large-superb-er")
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+ feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained("superb/hubert-large-superb-er")
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+
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+ # compute attention masks and normalize the waveform if needed
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+ inputs = feature_extractor(dataset[:4]["speech"], sampling_rate=16000, padding=True, return_tensors="pt")
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+
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+ logits = model(**inputs).logits
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+ predicted_ids = torch.argmax(logits, dim=-1)
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+ labels = [model.config.id2label[_id] for _id in predicted_ids.tolist()]
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+ ```
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+
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+ ## Eval results
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+
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+ The evaluation metric is accuracy.
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+
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+ | | **s3prl** | **transformers** |
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+ |--------|-----------|------------------|
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+ |**session1**| `0.6762` | `N/A` |
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+
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+ ### BibTeX entry and citation info
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+
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+ ```bibtex
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+ @article{yang2021superb,
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+ title={SUPERB: Speech processing Universal PERformance Benchmark},
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+ author={Yang, Shu-wen and Chi, Po-Han and Chuang, Yung-Sung and Lai, Cheng-I Jeff and Lakhotia, Kushal and Lin, Yist Y and Liu, Andy T and Shi, Jiatong and Chang, Xuankai and Lin, Guan-Ting and others},
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+ journal={arXiv preprint arXiv:2105.01051},
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+ year={2021}
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+ }
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+ ```
config.json ADDED
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+ {
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+ "_name_or_path": "facebook/hubert-large-ll60k",
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+ "activation_dropout": 0.0,
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+ "apply_spec_augment": true,
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+ "architectures": [
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+ "HubertForSequenceClassification"
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+ ],
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+ "attention_dropout": 0.1,
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+ "bos_token_id": 1,
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+ "classifier_proj_size": 256,
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+ "conv_bias": true,
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+ "conv_dim": [
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+ 512,
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+ 512,
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+ 512,
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+ 512,
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+ 512,
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+ 512,
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+ 512
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+ ],
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+ "conv_kernel": [
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+ 10,
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+ 3,
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+ 3,
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+ 3,
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+ 3,
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+ 2,
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+ 2
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+ ],
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+ "conv_stride": [
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+ 5,
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+ 2,
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+ 2,
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+ 2,
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+ 2,
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+ 2,
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+ 2
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+ ],
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+ "ctc_loss_reduction": "sum",
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+ "ctc_zero_infinity": false,
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+ "do_stable_layer_norm": true,
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+ "eos_token_id": 2,
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+ "feat_extract_activation": "gelu",
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+ "feat_extract_dropout": 0.0,
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+ "feat_extract_norm": "layer",
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+ "feat_proj_dropout": 0.1,
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+ "final_dropout": 0.0,
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+ "gradient_checkpointing": false,
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+ "hidden_act": "gelu",
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+ "hidden_dropout": 0.1,
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+ "hidden_size": 1024,
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+ "id2label": {
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+ "0": "neu",
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+ "1": "hap",
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+ "2": "ang",
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+ "3": "sad"
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+ },
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+ "initializer_range": 0.02,
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+ "intermediate_size": 4096,
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+ "label2id": {
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+ "ang": 2,
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+ "hap": 1,
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+ "neu": 0,
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+ "sad": 3
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+ },
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+ "layer_norm_eps": 1e-05,
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+ "layerdrop": 0.1,
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+ "mask_channel_length": 10,
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+ "mask_channel_min_space": 1,
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+ "mask_channel_other": 0.0,
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+ "mask_channel_prob": 0.0,
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+ "mask_channel_selection": "static",
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+ "mask_feature_length": 10,
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+ "mask_feature_prob": 0.0,
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+ "mask_time_length": 10,
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+ "mask_time_min_space": 1,
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+ "mask_time_other": 0.0,
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+ "mask_time_prob": 0.075,
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+ "mask_time_selection": "static",
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+ "model_type": "hubert",
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+ "num_attention_heads": 16,
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+ "num_conv_pos_embedding_groups": 16,
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+ "num_conv_pos_embeddings": 128,
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+ "num_feat_extract_layers": 7,
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+ "num_hidden_layers": 24,
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+ "pad_token_id": 0,
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+ "torch_dtype": "float32",
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+ "transformers_version": "4.11.0.dev0",
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+ "use_weighted_layer_sum": true,
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+ "vocab_size": 32
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+ }
preprocessor_config.json ADDED
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+ {
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+ "do_normalize": true,
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+ "feature_extractor_type": "Wav2Vec2FeatureExtractor",
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+ "feature_size": 1,
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+ "padding_side": "right",
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+ "padding_value": 0,
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+ "return_attention_mask": true,
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+ "sampling_rate": 16000
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+ }
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