Spaces:
Sleeping
Sleeping
Change output language to spanish.
Browse files
app.py
CHANGED
@@ -3,40 +3,38 @@ import numpy as np
|
|
3 |
import torch
|
4 |
from datasets import load_dataset
|
5 |
|
6 |
-
from transformers import
|
7 |
-
|
|
|
8 |
|
9 |
device = "cuda:0" if torch.cuda.is_available() else "cpu"
|
10 |
|
11 |
# load speech translation checkpoint
|
12 |
asr_pipe = pipeline("automatic-speech-recognition", model="openai/whisper-base", device=device)
|
13 |
|
14 |
-
# load text-to-speech checkpoint
|
15 |
-
|
16 |
-
|
17 |
-
model = SpeechT5ForTextToSpeech.from_pretrained("microsoft/speecht5_tts").to(device)
|
18 |
-
vocoder = SpeechT5HifiGan.from_pretrained("microsoft/speecht5_hifigan").to(device)
|
19 |
-
|
20 |
-
embeddings_dataset = load_dataset("Matthijs/cmu-arctic-xvectors", split="validation")
|
21 |
-
speaker_embeddings = torch.tensor(embeddings_dataset[7306]["xvector"]).unsqueeze(0)
|
22 |
|
23 |
|
24 |
def translate(audio):
|
25 |
-
outputs = asr_pipe(audio, max_new_tokens=256, generate_kwargs={"task": "
|
26 |
return outputs["text"]
|
27 |
|
28 |
|
29 |
def synthesise(text):
|
30 |
-
inputs =
|
31 |
-
|
32 |
-
|
|
|
|
|
33 |
|
34 |
|
35 |
def speech_to_speech_translation(audio):
|
36 |
translated_text = translate(audio)
|
37 |
synthesised_speech = synthesise(translated_text)
|
38 |
synthesised_speech = (synthesised_speech.numpy() * 32767).astype(np.int16)
|
39 |
-
return
|
40 |
|
41 |
|
42 |
title = "Cascaded STST"
|
@@ -51,7 +49,7 @@ demo = gr.Blocks()
|
|
51 |
|
52 |
mic_translate = gr.Interface(
|
53 |
fn=speech_to_speech_translation,
|
54 |
-
inputs=gr.Audio(
|
55 |
outputs=gr.Audio(label="Generated Speech", type="numpy"),
|
56 |
title=title,
|
57 |
description=description,
|
@@ -59,7 +57,7 @@ mic_translate = gr.Interface(
|
|
59 |
|
60 |
file_translate = gr.Interface(
|
61 |
fn=speech_to_speech_translation,
|
62 |
-
inputs=gr.Audio(
|
63 |
outputs=gr.Audio(label="Generated Speech", type="numpy"),
|
64 |
examples=[["./example.wav"]],
|
65 |
title=title,
|
|
|
3 |
import torch
|
4 |
from datasets import load_dataset
|
5 |
|
6 |
+
from transformers import pipeline
|
7 |
+
from transformers import VitsModel, AutoTokenizer
|
8 |
+
#import torch
|
9 |
|
10 |
device = "cuda:0" if torch.cuda.is_available() else "cpu"
|
11 |
|
12 |
# load speech translation checkpoint
|
13 |
asr_pipe = pipeline("automatic-speech-recognition", model="openai/whisper-base", device=device)
|
14 |
|
15 |
+
# load text-to-speech checkpoint
|
16 |
+
model = VitsModel.from_pretrained("facebook/mms-tts-spa")
|
17 |
+
tokenizer = AutoTokenizer.from_pretrained("facebook/mms-tts-spa")
|
|
|
|
|
|
|
|
|
|
|
18 |
|
19 |
|
20 |
def translate(audio):
|
21 |
+
outputs = asr_pipe(audio, max_new_tokens=256, generate_kwargs={"task": "transcribe", "language":"es"})
|
22 |
return outputs["text"]
|
23 |
|
24 |
|
25 |
def synthesise(text):
|
26 |
+
inputs = tokenizer(text, return_tensors="pt")
|
27 |
+
|
28 |
+
with torch.no_grad():
|
29 |
+
speech = model(**inputs).waveform
|
30 |
+
return speech[0].cpu()
|
31 |
|
32 |
|
33 |
def speech_to_speech_translation(audio):
|
34 |
translated_text = translate(audio)
|
35 |
synthesised_speech = synthesise(translated_text)
|
36 |
synthesised_speech = (synthesised_speech.numpy() * 32767).astype(np.int16)
|
37 |
+
return model.config.sampling_rate, synthesised_speech
|
38 |
|
39 |
|
40 |
title = "Cascaded STST"
|
|
|
49 |
|
50 |
mic_translate = gr.Interface(
|
51 |
fn=speech_to_speech_translation,
|
52 |
+
inputs=gr.Audio(sources=["microphone"], type="filepath"),
|
53 |
outputs=gr.Audio(label="Generated Speech", type="numpy"),
|
54 |
title=title,
|
55 |
description=description,
|
|
|
57 |
|
58 |
file_translate = gr.Interface(
|
59 |
fn=speech_to_speech_translation,
|
60 |
+
inputs=gr.Audio(sources=["upload"], type="filepath"),
|
61 |
outputs=gr.Audio(label="Generated Speech", type="numpy"),
|
62 |
examples=[["./example.wav"]],
|
63 |
title=title,
|