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import os | |
import torch | |
import librosa | |
import numpy as np | |
import gradio as gr | |
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor | |
def format_time(milliseconds): | |
seconds, milliseconds = divmod(int(milliseconds), 1000) | |
minutes, seconds = divmod(seconds, 60) | |
hours, minutes = divmod(minutes, 60) | |
return f"{hours:02d}:{minutes:02d}:{seconds:02d},{milliseconds:03d}" | |
def detect_speech_activity(y, sr, frame_length=1024, hop_length=512, threshold=0.01): | |
energy = librosa.feature.rms(y=y, frame_length=frame_length, hop_length=hop_length)[0] | |
speech_frames = energy > threshold | |
speech_regions = [] | |
in_speech = False | |
for i, speech in enumerate(speech_frames): | |
if speech and not in_speech: | |
start = i | |
in_speech = True | |
elif not speech and in_speech: | |
end = i | |
speech_regions.append((start * hop_length / sr, end * hop_length / sr)) | |
in_speech = False | |
if in_speech: | |
speech_regions.append((start * hop_length / sr, len(y) / sr)) | |
return speech_regions | |
def post_process_text(text): | |
text = text.replace(" ", " ") | |
text = text.strip() | |
return text | |
def transcribe_audio(audio_file): | |
device = torch.device('cuda' if torch.cuda.is_available() else 'cpu') | |
model_name = "Akashpb13/xlsr_kurmanji_kurdish" | |
processor = Wav2Vec2Processor.from_pretrained(model_name) | |
model = Wav2Vec2ForCTC.from_pretrained(model_name).to(device) | |
y, sr = librosa.load(audio_file, sr=16000) | |
voiced_segments = detect_speech_activity(y, sr, threshold=0.005) | |
srt_content = "" | |
for i, (start, end) in enumerate(voiced_segments, start=1): | |
segment_audio = y[int(start * sr):int(end * sr)] | |
input_values = processor(segment_audio, sampling_rate=sr, return_tensors="pt").input_values | |
input_values = input_values.to(device) | |
with torch.no_grad(): | |
logits = model(input_values).logits | |
predicted_ids = torch.argmax(logits, dim=-1) | |
transcription = processor.batch_decode(predicted_ids)[0] | |
transcription = post_process_text(transcription) | |
if transcription: | |
start_time = format_time(start * 1000) | |
end_time = format_time(end * 1000) | |
srt_content += f"{i}\n" | |
srt_content += f"{start_time} --> {end_time}\n" | |
# Break long lines into shorter ones (max 50 characters) | |
words = transcription.split() | |
lines = [] | |
current_line = "" | |
for word in words: | |
if len(current_line) + len(word) > 50: | |
lines.append(current_line.strip()) | |
current_line = "" | |
current_line += word + " " | |
if current_line: | |
lines.append(current_line.strip()) | |
srt_content += "\n".join(lines) + "\n\n" | |
return srt_content | |
def save_srt(audio_file): | |
srt_content = transcribe_audio(audio_file) | |
output_filename = "output.srt" | |
with open(output_filename, "w", encoding="utf-8") as f: | |
f.write(srt_content) | |
return output_filename, srt_content | |
iface = gr.Interface( | |
fn=save_srt, | |
inputs=gr.Audio(type="filepath"), | |
outputs=[ | |
gr.File(label="Download SRT"), | |
gr.Textbox(label="SRT Content", lines=10) | |
], | |
title="Kurdish Speech-to-Text Transcription", | |
description="Upload an audio file to generate a SRT subtitle file with Kurdish transcription." | |
) | |
if __name__ == "__main__": | |
iface.launch() |