# Install required packages import torch from transformers import AutoModelForSpeechSeq2Seq, AutoProcessor, pipeline from pydub import AudioSegment import os import gradio as gr # Load the model and processor model_id = "hackergeek98/tinyyyy_whisper" device = "cuda" if torch.cuda.is_available() else "cpu" model = AutoModelForSpeechSeq2Seq.from_pretrained(model_id).to(device) processor = AutoProcessor.from_pretrained(model_id) # Create pipeline whisper_pipe = pipeline( "automatic-speech-recognition", model=model, tokenizer=processor.tokenizer, feature_extractor=processor.feature_extractor, device=0 if torch.cuda.is_available() else -1 ) # Convert audio to WAV format def convert_to_wav(audio_path): audio = AudioSegment.from_file(audio_path) wav_path = "converted_audio.wav" audio.export(wav_path, format="wav") return wav_path # Split long audio into chunks def split_audio(audio_path, chunk_length_ms=30000): # Default: 30 sec per chunk audio = AudioSegment.from_wav(audio_path) chunks = [audio[i:i+chunk_length_ms] for i in range(0, len(audio), chunk_length_ms)] chunk_paths = [] for i, chunk in enumerate(chunks): chunk_path = f"chunk_{i}.wav" chunk.export(chunk_path, format="wav") chunk_paths.append(chunk_path) return chunk_paths # Transcribe a long audio file def transcribe_long_audio(audio_path): wav_path = convert_to_wav(audio_path) chunk_paths = split_audio(wav_path) transcription = "" for chunk in chunk_paths: result = whisper_pipe(chunk) transcription += result["text"] + "\n" os.remove(chunk) # Remove processed chunk os.remove(wav_path) # Cleanup original file return transcription # Gradio interface def transcribe_interface(audio_file): return transcribe_long_audio(audio_file) iface = gr.Interface( fn=transcribe_interface, inputs=gr.Audio(source="upload", type="filepath"), outputs="text", title="Whisper ASR - Transcription", description="Upload an audio file, and the model will transcribe it." ) if __name__ == "__main__": iface.launch()