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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: | |
![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.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() | |