import gradio as gr from transformers import pipeline, WhisperProcessor, WhisperForConditionalGeneration import torch import librosa import subprocess from langdetect import detect_langs import os import warnings from transformers import logging import math import json # Suppress warnings warnings.filterwarnings("ignore") logging.set_verbosity_error() # Updated models by language MODELS = { "es": [ "openai/whisper-large-v3", "facebook/wav2vec2-large-xlsr-53-spanish", "jonatasgrosman/wav2vec2-xls-r-1b-spanish" ], "en": [ "openai/whisper-large-v3", "facebook/wav2vec2-large-960h", "microsoft/wav2vec2-base-960h" ], "pt": [ "facebook/wav2vec2-large-xlsr-53-portuguese", "openai/whisper-medium", "jonatasgrosman/wav2vec2-large-xlsr-53-portuguese" ] } def convert_audio_to_wav(audio_path): if os.path.isdir(audio_path): raise ValueError(f"The path provided is a directory: {audio_path}") wav_path = "converted_audio.wav" command = ["ffmpeg", "-i", audio_path, "-ac", "1", "-ar", "16000", wav_path] subprocess.run(command, stdout=subprocess.PIPE, stderr=subprocess.PIPE) return wav_path def detect_language(audio_path): try: speech, _ = librosa.load(audio_path, sr=16000, duration=30) except Exception as e: raise ValueError(f"Error loading audio file with librosa: {e}") processor = WhisperProcessor.from_pretrained("openai/whisper-base") model = WhisperForConditionalGeneration.from_pretrained("openai/whisper-base") input_features = processor(speech, sampling_rate=16000, return_tensors="pt").input_features predicted_ids = model.generate(input_features) transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0] langs = detect_langs(transcription) es_confidence = next((lang.prob for lang in langs if lang.lang == 'es'), 0) pt_confidence = next((lang.prob for lang in langs if lang.lang == 'pt'), 0) if abs(es_confidence - pt_confidence) < 0.2: return 'es' return max(langs, key=lambda x: x.prob).lang def transcribe_audio_stream(audio, model_name): wav_audio = convert_audio_to_wav(audio) speech, rate = librosa.load(wav_audio, sr=16000) duration = len(speech) / rate transcriptions = [] if "whisper" in model_name: processor = WhisperProcessor.from_pretrained(model_name) model = WhisperForConditionalGeneration.from_pretrained(model_name) chunk_duration = 30 # seconds for i in range(0, int(duration), chunk_duration): end = min(i + chunk_duration, duration) chunk = speech[int(i * rate):int(end * rate)] input_features = processor(chunk, sampling_rate=16000, return_tensors="pt").input_features predicted_ids = model.generate(input_features) transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0] progress = min(100, (end / duration) * 100) transcriptions.append({ "start_time": i, "end_time": end, "text": transcription }) yield transcriptions, progress else: transcriber = pipeline("automatic-speech-recognition", model=model_name) chunk_duration = 10 # seconds for i in range(0, int(duration), chunk_duration): end = min(i + chunk_duration, duration) chunk = speech[int(i * rate):int(end * rate)] result = transcriber(chunk) progress = min(100, (end / duration) * 100) transcriptions.append({ "start_time": i, "end_time": end, "text": result["text"] }) yield transcriptions, progress def detect_and_select_model(audio): wav_audio = convert_audio_to_wav(audio) language = detect_language(wav_audio) model_options = MODELS.get(language, MODELS["en"]) return language, model_options def save_transcription(transcriptions, file_format): if file_format == "JSON": file_path = "transcription.json" with open(file_path, 'w') as f: json.dump(transcriptions, f, ensure_ascii=False, indent=4) elif file_format == "TXT": file_path = "transcription.txt" with open(file_path, 'w') as f: for entry in transcriptions: f.write(f"{entry['start_time']},{entry['end_time']},{entry['text']}\n") return file_path def combined_interface(audio, file_format): try: language, model_options = detect_and_select_model(audio) selected_model = model_options[0] yield language, model_options, selected_model, "", 0, "Initializing..." transcriptions = [] for partial_transcriptions, progress in transcribe_audio_stream(audio, selected_model): transcriptions = partial_transcriptions full_transcription = " ".join([t["text"] for t in transcriptions]) progress_int = math.floor(progress) status = f"Transcribing... {progress_int}% complete" yield language, model_options, selected_model, full_transcription.strip(), progress_int, status # Save transcription file file_path = save_transcription(transcriptions, file_format) # Clean up temporary files os.remove("converted_audio.wav") yield language, model_options, selected_model, full_transcription.strip(), 100, f"Transcription complete! Download {file_path}", file_path except Exception as e: yield str(e), [], "", "An error occurred during processing.", 0, "Error", "" iface = gr.Interface( fn=combined_interface, inputs=[ gr.Audio(type="filepath"), gr.Radio(choices=["JSON", "TXT"], label="Choose output format") ], outputs=[ gr.Textbox(label="Detected Language"), gr.Dropdown(label="Available Models", choices=[]), gr.Textbox(label="Selected Model"), gr.Textbox(label="Transcription", lines=10), gr.Slider(minimum=0, maximum=100, label="Progress", interactive=False), gr.Textbox(label="Status"), gr.File(label="Download Transcription") ], title="Multilingual Audio Transcriber with Real-time Display and Progress Indicator", description="Upload an audio file to detect the language, select the transcription model, and get the transcription in real-time. Optimized for Spanish, English, and Portuguese.", live=True ) if __name__ == "__main__": iface.queue().launch()