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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 as transformers_logging |
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import math |
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import json |
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import tempfile |
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import logging |
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import concurrent.futures |
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logging.basicConfig(level=logging.INFO) |
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warnings.filterwarnings("ignore") |
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transformers_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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"fr": [ |
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"jonatasgrosman/wav2vec2-large-xlsr-53-french" |
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] |
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} |
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model_cache = {} |
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def get_model(model_name): |
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if model_name not in model_cache: |
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model_cache[model_name] = WhisperForConditionalGeneration.from_pretrained(model_name) |
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return model_cache[model_name] |
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def verify_ffmpeg_installation(): |
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try: |
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subprocess.run(["ffmpeg", "-version"], stdout=subprocess.PIPE, stderr=subprocess.PIPE, check=True) |
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except subprocess.CalledProcessError as e: |
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logging.error("ffmpeg no est谩 instalado o no se puede ejecutar correctamente.") |
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raise e |
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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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with tempfile.NamedTemporaryFile(suffix=".wav", delete=False) as tmp: |
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wav_path = tmp.name |
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command = ["ffmpeg", "-y", "-i", audio_path, "-ac", "1", "-ar", "16000", wav_path] |
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process = subprocess.run(command, stdout=subprocess.PIPE, stderr=subprocess.PIPE) |
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logging.info(process.stdout.decode()) |
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logging.error(process.stderr.decode()) |
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if process.returncode != 0: |
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raise ValueError(f"Error al convertir el archivo de audio a wav: {process.stderr.decode()}") |
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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 = get_model("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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processor = WhisperProcessor.from_pretrained(model_name) |
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model = get_model(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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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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with tempfile.NamedTemporaryFile(delete=False, suffix=".json") as tmp: |
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json.dump(transcriptions, tmp, ensure_ascii=False, indent=4) |
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file_path = tmp.name |
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elif file_format == "TXT": |
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with tempfile.NamedTemporaryFile(delete=False, suffix=".txt") as tmp: |
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for entry in transcriptions: |
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tmp.write(f"{entry['start_time']:.2f},{entry['end_time']:.2f},{entry['text']}\n".encode()) |
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file_path = tmp.name |
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logging.info(f"Archivo de transcripci贸n guardado en: {file_path}") |
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return file_path |
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def combined_interface(audio, file_format, confirmed_language, chosen_model): |
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try: |
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logging.info(f"Ruta del archivo de audio subido: {audio}") |
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verify_ffmpeg_installation() |
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language, model_options = detect_and_select_model(audio) |
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if not confirmed_language: |
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confirmed_language = language |
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if not chosen_model: |
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chosen_model = model_options[0] |
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logging.info(f"Idioma detectado: {confirmed_language}") |
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logging.info(f"Modelos disponibles: {model_options}") |
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logging.info(f"Modelo seleccionado: {chosen_model}") |
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yield confirmed_language, model_options, chosen_model, "", 0, "Initializing...", None |
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transcriptions = [] |
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for partial_transcriptions, progress in transcribe_audio_stream(audio, chosen_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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logging.info(f"Progreso: {progress_int}%") |
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yield confirmed_language, model_options, chosen_model, full_transcription.strip(), progress_int, status, None |
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logging.info("Guardando transcripci贸n.") |
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file_path = save_transcription(transcriptions, file_format) |
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if os.path.isdir(file_path): |
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raise ValueError(f"El archivo de transcripci贸n deber铆a ser un archivo, pero es un directorio: {file_path}") |
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if not os.path.isfile(file_path): |
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raise ValueError(f"El archivo de transcripci贸n no existe: {file_path}") |
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os.remove("converted_audio.wav") |
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logging.info("Archivos temporales limpiados.") |
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yield confirmed_language, model_options, chosen_model, full_transcription.strip(), 100, "Transcription complete! Download the file below.", file_path |
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except Exception as e: |
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logging.error(f"Error: {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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gr.Dropdown(choices=["", "es", "en", "pt", "fr"], label="Confirm detected language (optional)"), |
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gr.Dropdown(choices=["", "openai/whisper-large-v3", "facebook/wav2vec2-large-xlsr-53-spanish", |
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"jonatasgrosman/wav2vec2-xls-r-1b-spanish", "microsoft/wav2vec2-base-960h"], label="Choose model (optional)") |
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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, confirm the detection or choose a 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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