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# import gradio as gr
# import numpy as np
# import torch
# from datasets import load_dataset
# 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-tiny", device=device)
# # load text-to-speech checkpoint and speaker embeddings
# model_id = "microsoft/speecht5_tts" #"Ellight/speecht5_finetuned_voxpopuli_nl" # update with your model id
# # pipe = pipeline("automatic-speech-recognition", model=model_id)
# model = SpeechT5ForTextToSpeech.from_pretrained(model_id)
# vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan")
# embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation",trust_remote_code=True))
# speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0)
# # speaker_embeddings = torch.tensor(embeddings_dataset[7440]["xvector"]).unsqueeze(0)
# processor = SpeechT5Processor.from_pretrained(model_id)
# replacements = [
# ("à", "a"),
# ("ç", "c"),
# ("è", "e"),
# ("ë", "e"),
# ("í", "i"),
# ("ï", "i"),
# ("ö", "o"),
# ("ü", "u"),
# ]
# def cleanup_text(text):
# for src, dst in replacements:
# text = text.replace(src, dst)
# return text
# def synthesize_speech(text):
# text = cleanup_text(text)
# inputs = processor(text=text, return_tensors="pt")
# speech = model.generate_speech(inputs["input_ids"].to(device), speaker_embeddings.to(device), vocoder=vocoder)
# return gr.Audio.update(value=(16000, speech.cpu().numpy()))
# def translate(audio):
# outputs = asr_pipe(audio, max_new_tokens=256, generate_kwargs={"task": "transcribe", "language": "Dutch"})
# return outputs["text"]
# def synthesise(text):
# text = cleanup_text(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
import gradio as gr
import numpy as np
import torch
from datasets import load_dataset
from transformers import SpeechT5ForTextToSpeech, SpeechT5HifiGan, SpeechT5Processor, pipeline
from transformers import VitsModel, VitsTokenizer
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("sanchit-gandhi/speecht5_tts_vox_nl").to(device)
vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device)
model = VitsModel.from_pretrained("Matthijs/mms-tts-nld")
tokenizer = VitsTokenizer.from_pretrained("Matthijs/mms-tts-nld")
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": "nl"})
return outputs["text"]
def synthesise(text):
inputs = tokenizer(text, return_tensors="pt")
with torch.no_grad():
outputs = model(inputs["input_ids"])
speech = outputs.audio[0]
return speech.cpu()
# def synthesise(text):
# inputs = processor(text=text, return_tensors="pt", padding='max_length', truncation=True)
# 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 = "Cascaded STST"
description = """
Demo for cascaded speech-to-speech translation (STST), mapping from source speech in any language to target speech in Dutch. Demo uses OpenAI's [Whisper Large v2](https://huggingface.co/openai/whisper-large-v2) model for speech translation, and [Sandiago21/speecht5_finetuned_voxpopuli_it](https://huggingface.co/Sandiago21/speecht5_finetuned_voxpopuli_it) checkpoint for text-to-speech, which is based on Microsoft's
[SpeechT5 TTS](https://huggingface.co/microsoft/speecht5_tts) model for text-to-speech, fine-tuned in Dutch Audio dataset:
![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() |