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import gradio as gr |
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from transformers import pipeline, WhisperProcessor, WhisperForConditionalGeneration |
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import torch |
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import librosa |
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import subprocess |
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from langdetect import detect_langs |
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import os |
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import warnings |
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from transformers import logging |
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import math |
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import json |
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warnings.filterwarnings("ignore") |
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logging.set_verbosity_error() |
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MODELS = { |
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"es": [ |
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"openai/whisper-large-v3", |
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"facebook/wav2vec2-large-xlsr-53-spanish", |
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"jonatasgrosman/wav2vec2-xls-r-1b-spanish" |
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], |
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"en": [ |
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"openai/whisper-large-v3", |
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"facebook/wav2vec2-large-960h", |
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"microsoft/wav2vec2-base-960h" |
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], |
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"pt": [ |
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"facebook/wav2vec2-large-xlsr-53-portuguese", |
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"openai/whisper-medium", |
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"jonatasgrosman/wav2vec2-xlsr-53-portuguese" |
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] |
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} |
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def convert_audio_to_wav(audio_path): |
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if os.path.isdir(audio_path): |
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raise ValueError(f"La ruta proporcionada es un directorio, no un archivo: {audio_path}") |
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wav_path = "converted_audio.wav" |
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command = ["ffmpeg", "-i", audio_path, "-ac", "1", "-ar", "16000", wav_path] |
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subprocess.run(command, stdout=subprocess.PIPE, stderr=subprocess.PIPE, check=True) |
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return wav_path |
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def detect_language(audio_path): |
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try: |
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speech, _ = librosa.load(audio_path, sr=16000, duration=30) |
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except Exception as e: |
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raise ValueError(f"Error al cargar el archivo de audio con librosa: {e}") |
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processor = WhisperProcessor.from_pretrained("openai/whisper-base") |
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model = WhisperForConditionalGeneration.from_pretrained("openai/whisper-base") |
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input_features = processor(speech, sampling_rate=16000, return_tensors="pt").input_features |
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predicted_ids = model.generate(input_features) |
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transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0] |
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langs = detect_langs(transcription) |
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es_confidence = next((lang.prob for lang in langs if lang.lang == 'es'), 0) |
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pt_confidence = next((lang.prob for lang in langs if lang.lang == 'pt'), 0) |
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if abs(es_confidence - pt_confidence) < 0.2: |
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return 'es' |
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return max(langs, key=lambda x: x.prob).lang |
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def transcribe_audio_stream(audio, model_name): |
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wav_audio = convert_audio_to_wav(audio) |
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speech, rate = librosa.load(wav_audio, sr=16000) |
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duration = len(speech) / rate |
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transcriptions = [] |
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if "whisper" in model_name: |
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processor = WhisperProcessor.from_pretrained(model_name) |
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model = WhisperForConditionalGeneration.from_pretrained(model_name) |
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chunk_duration = 30 |
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for i in range(0, int(duration), chunk_duration): |
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end = min(i + chunk_duration, duration) |
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chunk = speech[int(i * rate):int(end * rate)] |
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input_features = processor(chunk, sampling_rate=16000, return_tensors="pt").input_features |
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predicted_ids = model.generate(input_features) |
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transcription = processor.batch_decode(predicted_ids, skip_special_tokens=True)[0] |
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progress = min(100, (end / duration) * 100) |
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transcriptions.append({ |
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"start_time": i, |
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"end_time": end, |
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"text": transcription |
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}) |
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yield transcriptions, progress |
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else: |
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transcriber = pipeline("automatic-speech-recognition", model=model_name) |
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chunk_duration = 10 |
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for i in range(0, int(duration), chunk_duration): |
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end = min(i + chunk_duration, duration) |
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chunk = speech[int(i * rate):int(end * rate)] |
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result = transcriber(chunk) |
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progress = min(100, (end / duration) * 100) |
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transcriptions.append({ |
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"start_time": i, |
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"end_time": end, |
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"text": result["text"] |
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}) |
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yield transcriptions, progress |
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def detect_and_select_model(audio): |
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wav_audio = convert_audio_to_wav(audio) |
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language = detect_language(wav_audio) |
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model_options = MODELS.get(language, MODELS["en"]) |
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return language, model_options |
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def save_transcription(transcriptions, file_format): |
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if file_format == "JSON": |
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file_path = "transcription.json" |
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with open(file_path, 'w') as f: |
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json.dump(transcriptions, f, ensure_ascii=False, indent=4) |
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elif file_format == "TXT": |
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file_path = "transcription.txt" |
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with open(file_path, 'w') as f: |
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for entry in transcriptions: |
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f.write(f"{entry['start_time']:.2f},{entry['end_time']:.2f},{entry['text']}\n") |
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return file_path |
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def combined_interface(audio, file_format): |
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try: |
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language, model_options = detect_and_select_model(audio) |
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selected_model = model_options[0] |
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yield language, model_options, selected_model, "", 0, "Initializing...", None |
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transcriptions = [] |
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for partial_transcriptions, progress in transcribe_audio_stream(audio, selected_model): |
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transcriptions = partial_transcriptions |
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full_transcription = " ".join([t["text"] for t in transcriptions]) |
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progress_int = math.floor(progress) |
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status = f"Transcribing... {progress_int}% complete" |
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yield language, model_options, selected_model, full_transcription.strip(), progress_int, status, None |
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file_path = save_transcription(transcriptions, file_format) |
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os.remove("converted_audio.wav") |
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yield language, model_options, selected_model, full_transcription.strip(), 100, f"Transcription complete! Download {file_path}", file_path |
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except Exception as e: |
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yield str(e), [], "", "An error occurred during processing.", 0, "Error", "" |
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iface = gr.Interface( |
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fn=combined_interface, |
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inputs=[ |
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gr.Audio(type="filepath", label="Upload Audio File"), |
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gr.Radio(choices=["JSON", "TXT"], label="Choose output format") |
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], |
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outputs=[ |
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gr.Textbox(label="Detected Language"), |
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gr.Dropdown(label="Available Models", choices=[]), |
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gr.Textbox(label="Selected Model"), |
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gr.Textbox(label="Transcription", lines=10), |
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gr.Slider(minimum=0, maximum=100, label="Progress", interactive=False), |
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gr.Textbox(label="Status"), |
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gr.File(label="Download Transcription") |
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], |
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title="Multilingual Audio Transcriber with Real-time Display and Progress Indicator", |
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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.", |
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live=True |
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) |
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if __name__ == "__main__": |
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iface.queue().launch() |
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