Update app.py
Browse files
app.py
CHANGED
@@ -4,61 +4,69 @@ import numpy as np
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from transformers import Wav2Vec2ForSequenceClassification, Wav2Vec2FeatureExtractor
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import librosa
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#
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# Configuration
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#
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#
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num_labels = 8
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#
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#
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model = Wav2Vec2ForSequenceClassification.from_pretrained(
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feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained("facebook/wav2vec2-base-960h")
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#
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# Prediction Function
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#
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def predict(audio):
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if audio is None:
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return "No audio provided."
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# Gradio
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sample_rate, audio_data = audio
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#
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if sample_rate != 16000:
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audio_data = librosa.resample(
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#
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inputs = feature_extractor(audio_data, sampling_rate=16000, return_tensors="pt", padding=True)
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#
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model.eval()
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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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#
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predicted_level = pred_class + 3
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return f"Predicted L2 English Accuracy Level: {predicted_level}"
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#
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# Gradio Interface
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#
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iface = gr.Interface(
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fn=predict,
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inputs=gr.Audio(type="numpy", label="Record or Upload Audio"),
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outputs="text",
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title="
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description=(
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"This demo uses Wav2Vec2ForSequenceClassification
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"
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),
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allow_flagging="never"
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)
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from transformers import Wav2Vec2ForSequenceClassification, Wav2Vec2FeatureExtractor
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import librosa
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# --------------------------------------------------
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# Configuration
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# --------------------------------------------------
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# We have 3 classes: 0 = "low", 1 = "medium", 2 = "high"
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num_labels = 3
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# Load a base Wav2Vec2 model for classification with 3 labels.
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# The classification head will be randomly initialized.
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model = Wav2Vec2ForSequenceClassification.from_pretrained(
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"facebook/wav2vec2-base-960h",
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num_labels=num_labels
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)
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feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained("facebook/wav2vec2-base-960h")
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# Map integer predictions to textual labels
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label_map = {0: "low", 1: "medium", 2: "high"}
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# --------------------------------------------------
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# Prediction Function
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# --------------------------------------------------
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def predict(audio):
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if audio is None:
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return "No audio provided."
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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 not already
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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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# Extract features
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inputs = feature_extractor(audio_data, sampling_rate=16000, return_tensors="pt", padding=True)
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# Model inference
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model.eval()
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with torch.no_grad():
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logits = model(**inputs).logits
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# Argmax over logits -> integer class
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pred_class = torch.argmax(logits, dim=-1).item()
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# Convert integer class to textual label
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predicted_label = label_map.get(pred_class, "Unknown")
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return f"Predicted Level: {predicted_label}"
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# --------------------------------------------------
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# Gradio Interface
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# --------------------------------------------------
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iface = gr.Interface(
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fn=predict,
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inputs=gr.Audio(type="numpy", label="Record or Upload Audio"),
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outputs="text",
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title="3-Class Audio Classification Demo (Random)",
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description=(
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"This demo uses Wav2Vec2ForSequenceClassification with 3 classes (low, medium, high) "
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"but has not been fine-tuned, so the classification head is random. The predictions "
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"are not meaningful, but the pipeline demonstrates how a 3-class audio classifier can be set up."
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),
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allow_flagging="never"
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)
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