import gradio as gr import numpy as np import torch from datasets import load_dataset from transformers import pipeline, VitsModel, VitsTokenizer # from transformers import SpeechT5ForTextToSpeech, SpeechT5HifiGan, SpeechT5Processor, pipeline device = "cuda:0" if torch.cuda.is_available() else "cpu" # load speech translation checkpoint asr_pipe = pipeline("automatic-speech-recognition", model="openai/whisper-base", device=device) # load text-to-speech checkpoint and speaker embeddings # processor = SpeechT5Processor.from_pretrained("microsoft/speecht5_tts") # model = SpeechT5ForTextToSpeech.from_pretrained("microsoft/speecht5_tts").to(device) # vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device) # embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation") # speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0) # using VITS MMS TTS instead of T5 TTS model = VitsModel.from_pretrained("facebook/mms-tts-deu") tokenizer = VitsTokenizer.from_pretrained("facebook/mms-tts-deu") def translate(audio): try: outputs = asr_pipe(audio, generate_kwargs={"task": "translate", "return_timestamps": True}) return outputs["text"] except Exception as e: print(f"Error in translation: {e}") return "Error during translation" def synthesise(text): try: inputs = tokenizer(text, return_tensors="pt") input_ids = inputs["input_ids"] with torch.no_grad(): outputs = model(input_ids) speech = outputs["waveform"] speech = speech.cpu() return speech.squeeze() except Exception as e: print(f"Error in synthesis: {e}") return None def speech_to_speech_translation(audio): translated_text = translate(audio) print('translated text:\t', translated_text) if translated_text == "Error during translation": return None, None # Return None for both outputs in case of translation error. synthesised_speech = synthesise(translated_text) if synthesised_speech is None: return None, None # Return None for both outputs in case of synthesis error. synthesised_speech = (synthesised_speech.numpy() * 32767).astype(np.int16) return 16000, synthesised_speech title = "Cascaded STST" description = """ Demo for cascaded speech-to-speech translation (STST), mapping from source speech in any language to target speech in German. Demo uses OpenAI's [Whisper Base](https://huggingface.co/openai/whisper-base) model for speech translation, and Microsoft's [SpeechT5 TTS](https://huggingface.co/microsoft/speecht5_tts) model for text-to-speech: ![Cascaded STST](https://huggingface.co/datasets/huggingface-course/audio-course-images/resolve/main/s2st_cascaded.png "Diagram of cascaded speech to speech translation") """ demo = gr.Blocks() mic_translate = gr.Interface( fn=speech_to_speech_translation, inputs=gr.Microphone(type="filepath"), outputs=gr.Audio(label="Generated Speech", type="numpy"), title=title, description=description, ) file_translate = gr.Interface( fn=speech_to_speech_translation, inputs=gr.Audio(type="filepath"), outputs=gr.Audio(label="Generated Speech", type="numpy"), examples=[["./example.wav"]], title=title, description=description, ) with demo: gr.TabbedInterface([mic_translate, file_translate], ["Microphone", "Audio File"]) demo.launch(debug=True, height=600) # demo.launch(height=600)