openslr/librispeech_asr
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How to use sanchit-gandhi/wav2vec2-2-gpt2-regularisation with Transformers:
# Use a pipeline as a high-level helper
from transformers import pipeline
pipe = pipeline("automatic-speech-recognition", model="sanchit-gandhi/wav2vec2-2-gpt2-regularisation") # Load model directly
from transformers import AutoTokenizer, AutoModelForSpeechSeq2Seq
tokenizer = AutoTokenizer.from_pretrained("sanchit-gandhi/wav2vec2-2-gpt2-regularisation")
model = AutoModelForSpeechSeq2Seq.from_pretrained("sanchit-gandhi/wav2vec2-2-gpt2-regularisation", device_map="auto")YAML Metadata Error:"model-index[0].name" is not allowed to be empty
This model was trained from scratch on the librispeech_asr dataset. It achieves the following results on the evaluation set:
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The following hyperparameters were used during training:
| Training Loss | Epoch | Step | Validation Loss | Wer |
|---|---|---|---|---|
| 3.5506 | 2.8 | 2500 | 4.4928 | 1.8772 |
| 0.5145 | 5.61 | 5000 | 1.8942 | 1.1063 |
| 0.2736 | 8.41 | 7500 | 1.6550 | 1.0372 |
| 0.0807 | 11.21 | 10000 | 1.7601 | 1.0004 |
| 0.0439 | 14.01 | 12500 | 1.8014 | 1.0022 |
| 0.043 | 16.82 | 15000 | 1.8534 | 1.0097 |
| 0.0434 | 19.62 | 17500 | 1.8529 | 0.9977 |