File size: 4,841 Bytes
67beffe
 
 
 
 
 
 
 
0514036
2090f04
67beffe
 
 
dc94ed9
 
67beffe
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
4ea1ce4
 
 
67beffe
 
 
 
 
 
 
 
0514036
 
 
2090f04
 
67beffe
 
 
 
 
 
 
 
 
0514036
 
2090f04
 
 
67beffe
 
 
 
 
 
 
 
2090f04
 
0514036
67beffe
2090f04
 
4ea1ce4
2090f04
 
 
 
 
 
67beffe
 
2090f04
4ea1ce4
2090f04
 
 
 
67beffe
2090f04
 
 
67beffe
 
2090f04
67beffe
2090f04
67beffe
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
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
from random import sample
import gradio as gr
import torchaudio
import torch
import torch.nn as nn
import lightning_module
import pdb
import jiwer
from local.convert_metrics import nat2avaMOS, WER2INTELI
from local.indicator_plot import Intelligibility_Plot, Naturalness_Plot

# ASR part
from transformers import pipeline
p = pipeline("automatic-speech-recognition")

# WER part
transformation = jiwer.Compose([
    jiwer.ToLowerCase(),
    jiwer.RemoveWhiteSpace(replace_by_space=True),
    jiwer.RemoveMultipleSpaces(),
    jiwer.ReduceToListOfListOfWords(word_delimiter=" ")
])

# WPM part
from transformers import Wav2Vec2Processor, Wav2Vec2ForCTC
processor = Wav2Vec2Processor.from_pretrained("facebook/wav2vec2-xlsr-53-espeak-cv-ft")
phoneme_model = Wav2Vec2ForCTC.from_pretrained("facebook/wav2vec2-xlsr-53-espeak-cv-ft")
# phoneme_model =  pipeline(model="facebook/wav2vec2-xlsr-53-espeak-cv-ft") 
class ChangeSampleRate(nn.Module):
    def __init__(self, input_rate: int, output_rate: int):
        super().__init__()
        self.output_rate = output_rate
        self.input_rate = input_rate

    def forward(self, wav: torch.tensor) -> torch.tensor:
        # Only accepts 1-channel waveform input
        wav = wav.view(wav.size(0), -1)
        new_length = wav.size(-1) * self.output_rate // self.input_rate
        indices = (torch.arange(new_length) * (self.input_rate / self.output_rate))
        round_down = wav[:, indices.long()]
        round_up = wav[:, (indices.long() + 1).clamp(max=wav.size(-1) - 1)]
        output = round_down * (1. - indices.fmod(1.)).unsqueeze(0) + round_up * indices.fmod(1.).unsqueeze(0)
        return output

model = lightning_module.BaselineLightningModule.load_from_checkpoint("epoch=3-step=7459.ckpt").eval()

def calc_mos(audio_path, ref):
    wav, sr = torchaudio.load(audio_path, channels_first=True)  
    if wav.shape[0] > 1:
        wav = wav.mean(dim=0, keepdim=True) # Mono channel
    osr = 16_000
    batch = wav.unsqueeze(0).repeat(10, 1, 1)
    csr = ChangeSampleRate(sr, osr)
    out_wavs = csr(wav)
    # ASR
    trans = p(audio_path)["text"]
    # WER
    wer = jiwer.wer(ref, trans, truth_transform=transformation, hypothesis_transform=transformation)
    # WER convert to Intellibility score
    INTELI_score = WER2INTELI(wer*100)
    
    INT_fig = Intelligibility_Plot(INTELI_score)
    
    # MOS
    batch = {
        'wav': out_wavs,
        'domains': torch.tensor([0]),
        'judge_id': torch.tensor([288])
    }
    with torch.no_grad():
        output = model(batch)
    predic_mos = output.mean(dim=1).squeeze().detach().numpy()*2 + 3
    # MOS to AVA MOS
    AVA_MOS = nat2avaMOS(predic_mos)
    
    MOS_fig = Naturalness_Plot(AVA_MOS)
    
    # Phonemes per minute (PPM)
    with torch.no_grad():
        logits = phoneme_model(out_wavs).logits
    phone_predicted_ids = torch.argmax(logits, dim=-1)
    phone_transcription = processor.batch_decode(phone_predicted_ids)
    lst_phonemes = phone_transcription[0].split(" ")
    wav_vad = torchaudio.functional.vad(wav, sample_rate=sr)
    ppm = len(lst_phonemes) / (wav_vad.shape[-1] / sr) * 60
    # pdb.set_trace()
    return AVA_MOS, MOS_fig, INTELI_score, INT_fig, trans, phone_transcription, ppm


with open("local/description.md") as f:
    description = f.read()

# calc_mos("audio_2023-11-01_15-57-39.wav", "hello world")
# pdb.set_trace()
examples = [
    [None, "Once upon a time, there was a young rat named Arthur who couldn't make up his mind."],
    [None, "Whenever the other rats asked Arthur if he wanted to go to the park, he would say, 'I don't know.'"],
]
iface = gr.Interface(
  fn=calc_mos,
  inputs=[gr.Audio(type='filepath', label="Audio to evaluate"), 
          gr.Textbox(placeholder="Input reference here (Don't keep this empty)", label="Reference")],
  outputs=[gr.Textbox(placeholder="Naturalness Score, ranged from 0 to 5, the higher the better.", label="Naturalness Score, ranged from 0 to 5, the higher the better.", visible=False),
           gr.Plot(label="Naturalness Score, ranged from 0 to 5, the higher the better.", show_label=True, container=True),
           gr.Textbox(placeholder="Intelligibility Score", label = "Intelligibility Score, range from 0 to 100, the higher the better", visible=False),
           gr.Plot(label="Intelligibility Score, range from 0 to 100, the higher the better", show_label=True, container=True),
           gr.Textbox(placeholder="Hypothesis", label="Hypothesis"),
           gr.Textbox(placeholder="Predicted Phonemes", label="Predicted Phonemes", visible=False),
           gr.Textbox(placeholder="Speaking Rate, Phonemes per minutes", label="Speaking Rate, Phonemes per minutes", visible=False)],
  title="Speech Analysis by Laronix AI",
  description=description,
  allow_flagging="auto",
  examples=examples,
)

iface.launch()