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Update README.md

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  1. README.md +2 -3
README.md CHANGED
@@ -1,7 +1,6 @@
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  ---
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  language:
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  - en
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- datasets:
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  tags:
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  - speech
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  ---
@@ -33,13 +32,13 @@ The model is fine-tuned on the [LibriMix dataset](https://github.com/JorisCos/Li
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  # Usage
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  ## Speaker Diarization
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  ```python
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- from transformers import Wav2Vec2FeatureExtractor, UniSpeechSatForAudioFrameClassification
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  from datasets import load_dataset
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  import torch
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  dataset = load_dataset("hf-internal-testing/librispeech_asr_demo", "clean", split="validation")
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  feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained('microsoft/wavlm-base-plus-sd')
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- model = UniSpeechSatForAudioFrameClassification.from_pretrained('microsoft/wavlm-base-plus-sd')
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  # audio file is decoded on the fly
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  inputs = feature_extractor(dataset[0]["audio"]["array"], return_tensors="pt")
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  ---
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  language:
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  - en
 
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  tags:
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  - speech
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  ---
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  # Usage
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  ## Speaker Diarization
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  ```python
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+ from transformers import Wav2Vec2FeatureExtractor, WavLMForAudioFrameClassification
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  from datasets import load_dataset
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  import torch
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  dataset = load_dataset("hf-internal-testing/librispeech_asr_demo", "clean", split="validation")
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  feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained('microsoft/wavlm-base-plus-sd')
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+ model = WavLMForAudioFrameClassification.from_pretrained('microsoft/wavlm-base-plus-sd')
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  # audio file is decoded on the fly
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  inputs = feature_extractor(dataset[0]["audio"]["array"], return_tensors="pt")