Create app.py
Browse files
app.py
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import tempfile
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import shutil
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import numpy as np
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import librosa
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from fastapi import FastAPI, UploadFile, File, Form
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import uvicorn
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import gradio as gr
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from sklearn.preprocessing import StandardScaler
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app = FastAPI()
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CHUNK_DURATION = 30 # seconds
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SIMILARITY_THRESHOLD = 0.75
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@app.get("/")
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def home():
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return {"message": "MFCC Speaker Diarization Server Running"}
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# ๐น Feature extraction
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def extract_features(y, sr):
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features = []
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mfcc = librosa.feature.mfcc(y=y, sr=sr, n_mfcc=13)
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features.extend(np.mean(mfcc, axis=1))
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spectral_centroid = librosa.feature.spectral_centroid(y=y, sr=sr)
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features.append(np.mean(spectral_centroid))
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spectral_bandwidth = librosa.feature.spectral_bandwidth(y=y, sr=sr)
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features.append(np.mean(spectral_bandwidth))
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zcr = librosa.feature.zero_crossing_rate(y)
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features.append(np.mean(zcr))
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rms = librosa.feature.rms(y=y)
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features.append(np.mean(rms))
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return np.array(features)
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# ๐น Split audio into small segments
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def split_audio(y, sr, frame_sec=1.0):
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frame_len = int(sr * frame_sec)
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segments = []
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times = []
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for i in range(0, len(y), frame_len):
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segment = y[i:i+frame_len]
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if len(segment) < frame_len:
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continue
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energy = np.mean(np.abs(segment))
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if energy > 0.01:
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segments.append(segment)
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times.append((i/sr, (i+frame_len)/sr))
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return segments, times
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# ๐น Core diarization logic (shared by API + UI)
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def process_audio_file(file_path):
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y, sr = librosa.load(file_path, sr=None)
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total_duration = len(y) / sr
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all_segments = []
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speaker_embeddings = []
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speaker_labels = []
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speaker_count = 0
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current_time = 0
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while current_time < total_duration:
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start_sample = int(current_time * sr)
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end_sample = int(min((current_time + CHUNK_DURATION) * sr, len(y)))
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chunk = y[start_sample:end_sample]
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segments, times = split_audio(chunk, sr)
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for seg, (start, end) in zip(segments, times):
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feat = extract_features(seg, sr)
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if len(speaker_embeddings) > 0:
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scaler = StandardScaler()
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X = np.vstack(speaker_embeddings + [feat])
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X = scaler.fit_transform(X)
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feat_norm = X[-1]
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existing = X[:-1]
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else:
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feat_norm = feat
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existing = []
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assigned = False
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for i, emb in enumerate(existing):
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similarity = np.dot(feat_norm, emb) / (
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np.linalg.norm(feat_norm) * np.linalg.norm(emb)
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)
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if similarity > SIMILARITY_THRESHOLD:
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speaker_id = speaker_labels[i]
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assigned = True
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break
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if not assigned:
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speaker_count += 1
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speaker_id = f"SPEAKER_{speaker_count}"
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speaker_embeddings.append(feat)
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speaker_labels.append(speaker_id)
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all_segments.append({
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"speaker": speaker_id,
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"start": round(current_time + start, 2),
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"end": round(current_time + end, 2)
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})
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current_time += CHUNK_DURATION
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return {"segments": all_segments}
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# ๐น FastAPI endpoint
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@app.post("/transcribe")
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async def transcribe(audio: UploadFile = File(...), lang: str = Form("en")):
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temp = tempfile.NamedTemporaryFile(delete=False, suffix=".wav")
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with temp as buffer:
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shutil.copyfileobj(audio.file, buffer)
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result = process_audio_file(temp.name)
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return result
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# ๐น Gradio UI function
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def gradio_process(audio_file):
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if audio_file is None:
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return {"error": "No file uploaded"}
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return process_audio_file(audio_file)
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# ๐น Build Gradio Interface
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gradio_ui = gr.Interface(
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fn=gradio_process,
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inputs=gr.Audio(type="filepath", label="Upload Audio"),
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outputs=gr.JSON(label="Speaker Segments"),
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title="Speaker Diarization (CPU)",
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description="Upload audio and get speaker labels with timestamps"
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)
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# ๐น Mount Gradio into FastAPI
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from fastapi.middleware.wsgi import WSGIMiddleware
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app.mount("/ui", WSGIMiddleware(gradio_ui))
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# ๐น Run
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if __name__ == "__main__":
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uvicorn.run(app, host="0.0.0.0", port=7860)
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