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| import os | |
| import torch | |
| import gradio as gr | |
| import numpy as np | |
| import torch | |
| from datasets import load_dataset, Audio | |
| from transformers import SpeechT5ForTextToSpeech, SpeechT5HifiGan, SpeechT5Processor, pipeline | |
| from speechbrain.pretrained import EncoderClassifier | |
| 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) | |
| model = SpeechT5ForTextToSpeech.from_pretrained( | |
| "JanLilan/speecht5_finetuned_openslr-slr69-cat" | |
| ).to(device) | |
| vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device) | |
| ###################################################################################### | |
| ################################## SPEAKER EMBEDDING ################################# | |
| ###################################################################################### | |
| # we will try to translate with this voice embedding... Let's see what happen. else: | |
| dataset = load_dataset("projecte-aina/openslr-slr69-ca-trimmed-denoised", split="train") | |
| dataset = dataset.cast_column("audio", Audio(sampling_rate=16000)) | |
| # LOAD | |
| spk_model_name = "speechbrain/spkrec-xvect-voxceleb" | |
| speaker_model = EncoderClassifier.from_hparams( | |
| source=spk_model_name, | |
| run_opts={"device": device}, | |
| savedir=os.path.join("/tmp", spk_model_name), | |
| ) | |
| def create_speaker_embedding(waveform): | |
| with torch.no_grad(): | |
| speaker_embeddings = speaker_model.encode_batch(torch.tensor(waveform)) | |
| speaker_embeddings = torch.nn.functional.normalize(speaker_embeddings, dim=2) | |
| speaker_embeddings = speaker_embeddings.squeeze().cpu().numpy() | |
| return speaker_embeddings | |
| # we must take one speaker embeding | |
| checkpoint = "microsoft/speecht5_tts" | |
| processor = SpeechT5Processor.from_pretrained(checkpoint) | |
| # function to embedd | |
| def prepare_dataset(example): | |
| audio = example["audio"] | |
| example = processor( | |
| text=example["transcription"], | |
| audio_target=audio["array"], | |
| sampling_rate=audio["sampling_rate"], | |
| return_attention_mask=False, | |
| ) | |
| # strip off the batch dimension | |
| example["labels"] = example["labels"][0] | |
| # use SpeechBrain to obtain x-vector | |
| example["speaker_embeddings"] = create_speaker_embedding(audio["array"]) | |
| return example | |
| processed_example = prepare_dataset(dataset[2]) | |
| speaker_embeddings = torch.tensor(processed_example["speaker_embeddings"]).unsqueeze(0) | |
| # embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation") | |
| # speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0) | |
| def translate(audio): | |
| outputs = asr_pipe(audio, max_new_tokens=256, generate_kwargs={"task": "transcribe", "language": "catalan"}) | |
| return outputs["text"] | |
| def synthesise(text): | |
| inputs = processor(text=text, return_tensors="pt") | |
| speech = model.generate_speech(inputs["input_ids"].to(device), speaker_embeddings.to(device), vocoder=vocoder) | |
| return speech.cpu() | |
| def speech_to_speech_translation(audio): | |
| translated_text = translate(audio) | |
| synthesised_speech = synthesise(translated_text) | |
| synthesised_speech = (synthesised_speech.numpy() * 32767).astype(np.int16) | |
| return 16000, synthesised_speech | |
| title = "Demo STST - Multilingual to Català Speech" | |
| description = """ | |
| Demo for cascaded speech-to-speech translation (STST), mapping from source speech in any language to target speech in Català. Demo uses OpenAI's [Whisper Base](https://huggingface.co/openai/whisper-base) model for speech translation to català, and Microsoft's | |
| [SpeechT5 TTS](https://huggingface.co/microsoft/speecht5_tts) model for text-to-speech fine-tuned on [projecte-aina/openslr-slr69-ca-trimmed-denoised](https://huggingface.co/datasets/projecte-aina/openslr-slr69-ca-trimmed-denoised). | |
| This demo can be improve updating it with [projecte-aina/tts-ca-coqui-vits-multispeaker](https://huggingface.co/projecte-aina/tts-ca-coqui-vits-multispeaker) model: | |
|  | |
| """ | |
| demo = gr.Blocks() | |
| mic_translate = gr.Interface( | |
| fn=speech_to_speech_translation, | |
| inputs=gr.Audio(source="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(source="upload", 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() | |