Update app.py
Browse files
app.py
CHANGED
@@ -11,7 +11,7 @@ from transformers import Wav2Vec2FeatureExtractor, Wav2Vec2ForSequenceClassifica
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# ------------------------------------------------
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model_name = "facebook/wav2vec2-base-960h"
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#
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model = Wav2Vec2ForSequenceClassification.from_pretrained(
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model_name,
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num_labels=8
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@@ -26,24 +26,24 @@ model.eval()
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def classify_accuracy(audio):
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"""
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Receives a tuple (sample_rate, data) from Gradio when type='numpy'.
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"""
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if audio is None:
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return "No audio provided."
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sample_rate, data = audio
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# Ensure
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if not isinstance(data, np.ndarray):
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data = np.array(data)
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# Resample
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target_sr = 16000
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if sample_rate != target_sr:
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data = librosa.resample(data, orig_sr=sample_rate, target_sr=target_sr)
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sample_rate = target_sr
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# Extract features
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inputs = feature_extractor(
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data,
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sampling_rate=sample_rate,
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@@ -51,14 +51,14 @@ def classify_accuracy(audio):
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padding=True
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)
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with torch.no_grad():
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outputs = model(**inputs)
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logits = outputs.logits
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predicted_id = torch.argmax(logits, dim=-1).item()
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# Map 0..7
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accuracy_level = predicted_id + 3
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return f"Predicted Accuracy Level: {accuracy_level}"
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# ------------------------------------------------
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@@ -66,15 +66,15 @@ def classify_accuracy(audio):
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# ------------------------------------------------
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title = "Speech Accuracy Classifier (Base Wav2Vec2)"
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description = (
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"
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"The
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"This demo simply illustrates how to attach a classification head."
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)
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demo = gr.Interface(
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fn=classify_accuracy,
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inputs=gr.Audio(source="
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outputs="
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title=title,
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description=description,
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allow_flagging="never"
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# ------------------------------------------------
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model_name = "facebook/wav2vec2-base-960h"
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# Specify num_labels=8 to create a random classification head on top.
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model = Wav2Vec2ForSequenceClassification.from_pretrained(
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model_name,
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num_labels=8
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def classify_accuracy(audio):
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"""
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Receives a tuple (sample_rate, data) from Gradio when type='numpy'.
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Resamples if needed, runs a forward pass, and returns a 'level'.
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"""
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if audio is None:
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return "No audio provided."
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sample_rate, data = audio
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# Ensure data is a NumPy array.
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if not isinstance(data, np.ndarray):
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data = np.array(data)
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# Resample to 16kHz if needed.
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target_sr = 16000
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if sample_rate != target_sr:
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data = librosa.resample(data, orig_sr=sample_rate, target_sr=target_sr)
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sample_rate = target_sr
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# Extract features from the audio data.
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inputs = feature_extractor(
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data,
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sampling_rate=sample_rate,
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padding=True
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)
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# Run model inference.
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with torch.no_grad():
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outputs = model(**inputs)
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logits = outputs.logits
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predicted_id = torch.argmax(logits, dim=-1).item()
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# Map predicted id (0..7) to the final level (3..10).
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accuracy_level = predicted_id + 3
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return f"Predicted Accuracy Level: {accuracy_level}"
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# ------------------------------------------------
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# ------------------------------------------------
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title = "Speech Accuracy Classifier (Base Wav2Vec2)"
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description = (
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"Record audio using your microphone or upload an audio file (left). "
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"The model (not fine-tuned) will classify the audio into an accuracy level (right)."
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)
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# Using source="microphone" allows for direct recording, while recent versions also enable file upload.
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demo = gr.Interface(
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fn=classify_accuracy,
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inputs=gr.Audio(source="microphone", type="numpy", label="Record/Upload Audio"),
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outputs=gr.Textbox(label="Classification Result"),
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title=title,
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description=description,
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allow_flagging="never"
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