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Create app.py
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app.py
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import gradio as gr
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import whisper
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import datetime
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import torch
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import subprocess
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from pyannote.audio import Audio
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from pyannote.audio.pipelines.speaker_verification import PretrainedSpeakerEmbedding
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from pyannote.core import Segment
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import wave
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import contextlib
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from sklearn.cluster import AgglomerativeClustering
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import numpy as np
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# Load Whisper model
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model_size = "medium.en"
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model = whisper.load_model(model_size)
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audio = Audio()
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embedding_model = PretrainedSpeakerEmbedding("speechbrain/spkrec-ecapa-voxceleb", device=torch.device("cuda" if torch.cuda.is_available() else "cpu"))
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def transcribe_and_diarize(audio_file, num_speakers=2):
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path = audio_file.name
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# Convert to WAV if necessary
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if path[-3:] != 'wav':
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subprocess.call(['ffmpeg', '-i', path, 'audio.wav', '-y'])
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path = 'audio.wav'
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# Transcribe audio
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result = model.transcribe(path)
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segments = result["segments"]
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# Get audio duration
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with contextlib.closing(wave.open(path, 'r')) as f:
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frames = f.getnframes()
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rate = f.getframerate()
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duration = frames / float(rate)
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# Define function to extract segment embeddings
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def segment_embedding(segment):
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start = segment["start"]
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end = min(duration, segment["end"])
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clip = Segment(start, end)
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waveform, sample_rate = audio.crop(path, clip)
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return embedding_model(waveform[None])
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# Extract embeddings for each segment
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embeddings = np.zeros(shape=(len(segments), 192))
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for i, segment in enumerate(segments):
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embeddings[i] = segment_embedding(segment)
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embeddings = np.nan_to_num(embeddings)
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# Perform speaker clustering
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clustering = AgglomerativeClustering(num_speakers).fit(embeddings)
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labels = clustering.labels_
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for i in range(len(segments)):
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segments[i]["speaker"] = 'SPEAKER ' + str(labels[i] + 1)
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# Generate transcript
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transcript = ""
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for i, segment in enumerate(segments):
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if i == 0 or segments[i - 1]["speaker"] != segment["speaker"]:
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transcript += "\n" + segment["speaker"] + ' ' + str(datetime.timedelta(seconds=round(segment["start"]))) + '\n'
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transcript += segment["text"][1:] + ' '
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transcript += "\n\n"
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return transcript
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iface = gr.Interface(
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fn=transcribe_and_diarize,
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inputs=[
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gr.Audio(source="upload", type="file"),
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gr.Number(value=2, label="Number of Speakers")
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],
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outputs="text",
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title="Audio Transcription and Speaker Diarization",
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description="Upload an audio file to get a transcription with speaker diarization."
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)
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iface.launch()
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