import torch from transformers import pipeline from transformers.pipelines.audio_utils import ffmpeg_read import gradio as gr MODEL_NAME = "EwoutLagendijk/whisper-small-indonesian" BATCH_SIZE = 8 device = 0 if torch.cuda.is_available() else "cpu" pipe = pipeline( task="automatic-speech-recognition", model=MODEL_NAME, chunk_length_s=30, device=device, ) # Copied from https://github.com/openai/whisper/blob/c09a7ae299c4c34c5839a76380ae407e7d785914/whisper/utils.py#L50 def format_timestamp(seconds: float, always_include_hours: bool = False, decimal_marker: str = "."): if seconds is not None: milliseconds = round(seconds * 1000.0) hours = milliseconds // 3_600_000 milliseconds -= hours * 3_600_000 minutes = milliseconds // 60_000 milliseconds -= minutes * 60_000 seconds = milliseconds // 1_000 milliseconds -= seconds * 1_000 hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else "" return f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}" else: # we have a malformed timestamp so just return it as is return seconds def transcribe_speech(filepath): # Load the audio audio, sampling_rate = librosa.load(filepath, sr=16000) # Define chunk size (e.g., 30 seconds) chunk_duration = 30 # in seconds chunk_samples = chunk_duration * sampling_rate # Process audio in chunks transcription = [] for i in range(0, len(audio), chunk_samples): chunk = audio[i:i + chunk_samples] # Convert the chunk into input features inputs = processor(audio=chunk, sampling_rate=16000, return_tensors="pt").input_features # Generate transcription for the chunk generated_ids = model.generate( inputs, max_new_tokens=444, # Max allowed by Whisper forced_decoder_ids=processor.get_decoder_prompt_ids(language="id", task="transcribe") ) # Decode and append the transcription chunk_transcription = processor.batch_decode(generated_ids, skip_special_tokens=True)[0] transcription.append(chunk_transcription) # Combine all chunk transcriptions into a single string return " ".join(transcription) demo = gr.Blocks() mic_transcribe = gr.Interface( fn=transcribe_speech, inputs=gr.Audio(sources="microphone", type="filepath"), outputs=gr.components.Textbox(), ) file_transcribe = gr.Interface( fn=transcribe_speech, inputs=gr.Audio(sources="upload", type="filepath"), outputs=gr.components.Textbox(), ) with demo: gr.TabbedInterface([mic_transcribe, file_transcribe], ["Transcribe Microphone", "Transcribe Audio File"]) demo.launch(share=True, debug=True)