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Browse files- app.py +136 -0
- requirements.txt +6 -0
app.py
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import gradio as gr
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import torch
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import torchaudio
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from transformers import AutoModelForCTC, AutoProcessor, pipeline
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from pydub import AudioSegment
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import numpy as np
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import librosa
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import io
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import tempfile
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# Load ASR model and processor for Hindi speech recognition
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print("Loading ASR model...")
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asr_processor = AutoProcessor.from_pretrained("ai4bharat/indicwav2vec-hindi")
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asr_model = AutoModelForCTC.from_pretrained("ai4bharat/indicwav2vec-hindi")
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# Load sentiment analysis pipeline for Hindi text
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print("Loading sentiment analysis model...")
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sentiment_pipeline = pipeline(
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"text-classification",
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model="LondonStory/txlm-roberta-hindi-sentiment",
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return_all_scores=True
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)
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# Move models to appropriate device (CPU for free Hugging Face Space)
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device = "cuda" if torch.cuda.is_available() else "cpu"
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asr_model.to(device)
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print(f"Models loaded on device: {device}")
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def predict(audio_filepath):
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"""
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Main prediction function that processes Hindi audio and returns sentiment analysis.
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Args:
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audio_filepath: Path to the uploaded audio file
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Returns:
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Dictionary with sentiment labels and confidence scores
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"""
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try:
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# Load and preprocess audio
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print(f"Processing audio file: {audio_filepath}")
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# Load audio using librosa and resample to 16kHz as required by the ASR model
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audio_array, sample_rate = librosa.load(audio_filepath, sr=16000)
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# Ensure audio is in the correct format
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if len(audio_array.shape) > 1:
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audio_array = np.mean(audio_array, axis=1)
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# Process audio with ASR processor
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inputs = asr_processor(
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audio_array,
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sampling_rate=16000,
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return_tensors="pt",
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padding=True
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)
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# Move inputs to device
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inputs = {k: v.to(device) for k, v in inputs.items()}
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# Transcribe audio to Hindi text
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with torch.no_grad():
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logits = asr_model(**inputs).logits
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# Get predicted token IDs
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predicted_ids = torch.argmax(logits, dim=-1)
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# Decode the transcription
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transcription = asr_processor.batch_decode(predicted_ids)[0]
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print(f"Transcribed text: {transcription}")
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# Handle empty transcription
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if not transcription.strip():
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return {"error": "Could not transcribe audio. Please ensure you're speaking in Hindi."}
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# Perform sentiment analysis on the transcribed text
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sentiment_results = sentiment_pipeline(transcription)
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# Format results for Gradio
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result_dict = {}
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for result in sentiment_results[0]:
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label = result['label']
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score = result['score']
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result_dict[label] = float(score)
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# Add transcription to results for reference
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result_dict['Transcription'] = transcription
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return result_dict
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except Exception as e:
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print(f"Error processing audio: {str(e)}")
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return {"error": f"Error processing audio: {str(e)}"}
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# Create Gradio interface
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demo = gr.Interface(
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fn=predict,
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inputs=gr.Audio(
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type="filepath",
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label="Upload Hindi Speech",
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sources=["upload", "microphone"]
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),
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outputs=gr.Label(
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label="Sentiment Analysis Result",
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num_top_classes=3
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),
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title="🎤 Hindi Speech Sentiment Analysis",
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description="""
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### Upload or record Hindi audio to analyze sentiment
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This app performs the following steps:
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1. **Speech Recognition**: Converts your Hindi speech to text using AI4Bharat's IndicWav2Vec model
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2. **Sentiment Analysis**: Analyzes the emotional tone using a specialized Hindi sentiment model
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**Instructions**:
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- Upload an audio file or record directly using the microphone
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- Speak clearly in Hindi for best results
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- The app will show both the transcribed text and sentiment scores
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**Supported sentiments**: Positive, Negative, and Neutral with confidence scores
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""",
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examples=None,
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theme=gr.themes.Soft(),
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allow_flagging="never"
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)
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# Launch the app
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if __name__ == "__main__":
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# Launch with share=True for public access, queue for handling multiple requests
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demo.launch(
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share=False, # Set to True if you want a public link for testing
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server_name="0.0.0.0", # Required for Hugging Face Spaces
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server_port=7860, # Default port for Hugging Face Spaces
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show_error=True
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)
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requirements.txt
ADDED
|
@@ -0,0 +1,6 @@
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| 1 |
+
transformers
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| 2 |
+
torch
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+
torchaudio
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| 4 |
+
datasets
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| 5 |
+
gradio
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+
pydub
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