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Update app.py
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app.py
CHANGED
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@@ -18,7 +18,7 @@ feature_extractor = Wav2Vec2FeatureExtractor.from_pretrained(MODEL_NAME)
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model = Wav2Vec2ForSequenceClassification.from_pretrained(MODEL_NAME).to(DEVICE)
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model.eval()
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# Use the model
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config = AutoConfig.from_pretrained(MODEL_NAME)
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LABELS = [config.id2label[i] for i in range(len(config.id2label))]
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@@ -40,55 +40,78 @@ def predict(audio):
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try:
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if audio is None:
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return {"Error": "No audio provided"}, None
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sr, data = audio
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data = np.array(data, dtype=np.float32)
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# Stereo -> Mono
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if len(data.shape) > 1:
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data = np.mean(data, axis=1)
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# Resample to 16kHz
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if sr != 16000:
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data = torchaudio.functional.resample(torch.tensor(data), sr, 16000).numpy()
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sr = 16000
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#
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inputs = feature_extractor(
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data,
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sampling_rate=sr,
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return_tensors="pt",
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padding=True
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)
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# Move tensors to device
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for k in inputs:
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inputs[k] = inputs[k].to(DEVICE)
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# Forward pass
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with torch.no_grad():
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logits = model(**inputs).logits
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probs = torch.nn.functional.softmax(logits, dim=-1)[0].cpu().numpy()
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#
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# Generate waveform plot
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fig, ax = plt.subplots(figsize=(
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ax.
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ax.
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ax.set_ylabel("Amplitude")
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ax.
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ax.set_yticks([])
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plt.tight_layout()
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return result, fig
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except Exception as e:
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return {"Error": str(e)}, None
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@@ -98,19 +121,28 @@ def predict(audio):
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demo = gr.Interface(
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fn=predict,
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inputs=gr.Audio(sources=["upload", "microphone"], type="numpy", label="🎤 Upload or Record Audio"),
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outputs=[
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description=(
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"Fine-tuned Wav2Vec2 model
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"
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"
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"
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),
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allow_flagging="never",
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)
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# =========================
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# LAUNCH
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# =========================
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if __name__ == "__main__":
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demo.launch()
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model = Wav2Vec2ForSequenceClassification.from_pretrained(MODEL_NAME).to(DEVICE)
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model.eval()
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# Use the model's label mapping directly
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config = AutoConfig.from_pretrained(MODEL_NAME)
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LABELS = [config.id2label[i] for i in range(len(config.id2label))]
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try:
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if audio is None:
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return {"Error": "No audio provided"}, None
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sr, data = audio
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data = np.array(data, dtype=np.float32)
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# Stereo -> Mono
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if len(data.shape) > 1:
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data = np.mean(data, axis=1)
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# Resample to 16kHz
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if sr != 16000:
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data = torchaudio.functional.resample(torch.tensor(data), sr, 16000).numpy()
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sr = 16000
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# Improved normalization - normalize to [-1, 1] range
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# Check if data is in int16 range or already normalized
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if np.abs(data).max() > 1.0:
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data = data / np.abs(data).max() # Normalize by max value
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# Apply gentle audio preprocessing to improve feature extraction
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# Remove DC offset
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data = data - np.mean(data)
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# Apply light pre-emphasis filter to balance frequencies
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pre_emphasis = 0.97
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data = np.append(data[0], data[1:] - pre_emphasis * data[:-1])
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# Feature extraction with proper padding
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inputs = feature_extractor(
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data,
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sampling_rate=sr,
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return_tensors="pt",
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padding=True,
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max_length=16000 * 10, # Max 10 seconds
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truncation=True
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)
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# Move tensors to device
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for k in inputs:
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inputs[k] = inputs[k].to(DEVICE)
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# Forward pass
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with torch.no_grad():
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logits = model(**inputs).logits
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# Apply temperature scaling to reduce overconfidence
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# Lower temperature = more uniform distribution
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temperature = 1.5
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logits = logits / temperature
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probs = torch.nn.functional.softmax(logits, dim=-1)[0].cpu().numpy()
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# Show ALL emotions with their scores (not just top 3)
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result = {}
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for i, label in enumerate(LABELS):
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emoji = EMOJIS.get(label, '')
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result[f"{label} {emoji}"] = round(float(probs[i]), 4)
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# Sort by probability
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result = dict(sorted(result.items(), key=lambda x: x[1], reverse=True))
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# Generate waveform plot
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fig, ax = plt.subplots(figsize=(8, 3))
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time_axis = np.linspace(0, len(data) / sr, len(data))
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ax.plot(time_axis, data, color='purple', linewidth=0.5)
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ax.set_title("Audio Waveform", fontsize=12, fontweight='bold')
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ax.set_xlabel("Time (seconds)")
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ax.set_ylabel("Amplitude")
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ax.grid(True, alpha=0.3)
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plt.tight_layout()
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return result, fig
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except Exception as e:
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return {"Error": str(e)}, None
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demo = gr.Interface(
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fn=predict,
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inputs=gr.Audio(sources=["upload", "microphone"], type="numpy", label="🎤 Upload or Record Audio"),
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outputs=[
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gr.Label(num_top_classes=7, label="Emotion Probabilities"),
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gr.Plot(label="Waveform Visualization")
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],
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title="🎧 Audio Emotion Detection",
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description=(
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"Fine-tuned Wav2Vec2 model for emotion recognition from voice. "
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"Detects: **Angry, Disgusted, Fearful, Happy, Neutral, Sad, Surprised**.\n\n"
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"**Tips for better results:**\n"
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"- Speak clearly and naturally\n"
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"- Record at least 2-3 seconds of audio\n"
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"- Avoid background noise\n"
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"- Try exaggerating emotions for testing\n\n"
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"Audio is automatically resampled to 16kHz and normalized."
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),
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examples=[],
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allow_flagging="never",
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theme=gr.themes.Soft()
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)
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# =========================
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# LAUNCH
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# =========================
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if __name__ == "__main__":
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demo.launch()
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