Update app.py
Browse files
app.py
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@@ -5,13 +5,10 @@ import librosa
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from transformers import Wav2Vec2ForSequenceClassification, Wav2Vec2FeatureExtractor
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# --------------------------------------------------
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#
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# --------------------------------------------------
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#
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# Load your fine-tuned model from the Hugging Face Hub
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# (Replace "Yilin0601/wav2vec2-accuracy-checkpoints" with your actual repo if different)
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model = Wav2Vec2ForSequenceClassification.from_pretrained(
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"Yilin0601/wav2vec2-accuracy-checkpoints"
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)
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@@ -29,11 +26,15 @@ def predict(audio):
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# Gradio provides audio as (sample_rate, np.array)
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sample_rate, audio_data = audio
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# Convert stereo to mono if needed
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if len(audio_data.shape) > 1 and audio_data.shape[1] > 1:
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audio_data = np.mean(audio_data, axis=1)
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# Resample to 16 kHz if
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if sample_rate != 16000:
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audio_data = librosa.resample(audio_data, orig_sr=sample_rate, target_sr=16000)
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@@ -50,11 +51,9 @@ def predict(audio):
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with torch.no_grad():
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logits = model(**inputs).logits
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#
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pred_class = torch.argmax(logits, dim=-1).item()
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# Map [0..7] back to levels [3..10] by adding 3
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predicted_level = pred_class + 3
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return f"Predicted Level: {predicted_level}"
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from transformers import Wav2Vec2ForSequenceClassification, Wav2Vec2FeatureExtractor
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# --------------------------------------------------
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# Load Your Fine-Tuned Model
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# --------------------------------------------------
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# This model was fine-tuned with labels remapped from [3..10] to [0..7].
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# Make sure the model repo name below is correct and accessible.
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model = Wav2Vec2ForSequenceClassification.from_pretrained(
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"Yilin0601/wav2vec2-accuracy-checkpoints"
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)
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# Gradio provides audio as (sample_rate, np.array)
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sample_rate, audio_data = audio
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# Ensure the audio is floating-point (librosa requires float32 or float64)
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if audio_data.dtype not in [np.float32, np.float64]:
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audio_data = audio_data.astype(np.float32)
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# Convert stereo to mono if needed
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if len(audio_data.shape) > 1 and audio_data.shape[1] > 1:
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audio_data = np.mean(audio_data, axis=1)
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# Resample to 16 kHz if needed
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if sample_rate != 16000:
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audio_data = librosa.resample(audio_data, orig_sr=sample_rate, target_sr=16000)
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with torch.no_grad():
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logits = model(**inputs).logits
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# The model output is an 8-class prediction (0..7), corresponding to original labels 3..10
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pred_class = torch.argmax(logits, dim=-1).item()
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predicted_level = pred_class + 3 # Map back to [3..10]
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return f"Predicted Level: {predicted_level}"
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