File size: 3,101 Bytes
d347764
ff3f816
d347764
 
 
9e2a87c
0c8ad01
 
d347764
 
 
 
0c8ad01
9e2a87c
d347764
e970d56
472a003
0c8ad01
d347764
9e2a87c
6ab6711
 
d347764
 
 
0c8ad01
 
 
bc20e42
e467863
 
 
be03dff
d347764
 
6ab6711
 
 
d614113
af4a9d1
d347764
 
 
 
 
 
 
 
 
 
f805e49
 
d386a01
 
f805e49
 
 
 
c737803
 
 
d347764
226ec3a
d347764
f805e49
 
d347764
c737803
 
 
 
 
 
 
 
 
 
 
3946ba6
c737803
49c1298
9e2a87c
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
import gradio as gr
import logging
import numpy as np
import torch

from transformers import VitsModel, VitsTokenizer, pipeline
from transformers import M2M100ForConditionalGeneration
from tokenization_small100 import SMALL100Tokenizer


device = "cuda:0" if torch.cuda.is_available() else "cpu"

target_language = "fr"

# load speech translation checkpoint
asr_pipe = pipeline("automatic-speech-recognition", model="bofenghuang/whisper-small-cv11-french", device=device)
translation_model = M2M100ForConditionalGeneration.from_pretrained("alirezamsh/small100")
translation_tokenizer = SMALL100Tokenizer.from_pretrained("alirezamsh/small100", tgt_lang=target_language)

# load text-to-speech checkpoint
model = VitsModel.from_pretrained("facebook/mms-tts-fra")
tokenizer = VitsTokenizer.from_pretrained("facebook/mms-tts-fra")


def translate(audio):
    outputs = asr_pipe(audio, max_new_tokens=256, generate_kwargs={"task": "translate"})
    eng_text = outputs["text"]
    encoded_eng_text = translation_tokenizer(eng_text, return_tensors="pt")
    generated_tokens = translation_model.generate(**encoded_eng_text)
    translated_text = translation_tokenizer.batch_decode(generated_tokens, skip_special_tokens=True)
    logging.info(f"Translated Text: {translated_text}")
    return translated_text


def synthesise(text):
    inputs = tokenizer(text, return_tensors="pt")
    with torch.no_grad():
        outputs = model(inputs["input_ids"])
    speech = outputs["waveform"][0]
    logging.info(speech)
    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 French. Demo uses OpenAI's [Whisper Base](https://huggingface.co/openai/whisper-base) model for ASR, the 
[SMaLL-100](https://huggingface.co/alirezamsh/small100) model for text to text translation and Facebook's[MMS TTS-FRA](https://huggingface.co/facebook/mms-tts-fra) for text-to-speech for french:

![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"])

logging.getLogger().setLevel(logging.INFO)
demo.launch()