Spaces:
Sleeping
Sleeping
Update app.py
Browse files
app.py
CHANGED
@@ -1,179 +1,18 @@
|
|
1 |
-
import
|
2 |
-
import
|
3 |
-
|
4 |
-
from typing import List, Optional, Union, Dict
|
5 |
-
|
6 |
-
import tqdm
|
7 |
import torch
|
8 |
-
import torchaudio
|
9 |
-
import numpy as np
|
10 |
-
import pandas as pd
|
11 |
-
from torch import nn
|
12 |
-
from torch.utils.data import DataLoader
|
13 |
-
from torch.nn import functional as F
|
14 |
-
from transformers import (
|
15 |
-
AutoFeatureExtractor,
|
16 |
-
AutoModelForAudioClassification,
|
17 |
-
Wav2Vec2Processor
|
18 |
-
)
|
19 |
-
|
20 |
-
class CustomDataset(torch.utils.data.Dataset):
|
21 |
-
def __init__(
|
22 |
-
self,
|
23 |
-
dataset: List,
|
24 |
-
basedir: Optional[str] = None,
|
25 |
-
sampling_rate: int = 16000,
|
26 |
-
max_audio_len: int = 5,
|
27 |
-
):
|
28 |
-
self.dataset = dataset
|
29 |
-
self.basedir = basedir
|
30 |
-
|
31 |
-
self.sampling_rate = sampling_rate
|
32 |
-
self.max_audio_len = max_audio_len
|
33 |
-
|
34 |
-
def __len__(self):
|
35 |
-
"""
|
36 |
-
Return the length of the dataset
|
37 |
-
"""
|
38 |
-
return len(self.dataset)
|
39 |
-
|
40 |
-
def __getitem__(self, index):
|
41 |
-
if self.basedir is None:
|
42 |
-
filepath = self.dataset[index]
|
43 |
-
else:
|
44 |
-
filepath = os.path.join(self.basedir, self.dataset[index])
|
45 |
-
|
46 |
-
speech_array, sr = torchaudio.load(filepath)
|
47 |
-
|
48 |
-
if speech_array.shape[0] > 1:
|
49 |
-
speech_array = torch.mean(speech_array, dim=0, keepdim=True)
|
50 |
-
|
51 |
-
if sr != self.sampling_rate:
|
52 |
-
transform = torchaudio.transforms.Resample(sr, self.sampling_rate)
|
53 |
-
speech_array = transform(speech_array)
|
54 |
-
sr = self.sampling_rate
|
55 |
-
|
56 |
-
len_audio = speech_array.shape[1]
|
57 |
-
|
58 |
-
# Pad or truncate the audio to match the desired length
|
59 |
-
if len_audio < self.max_audio_len * self.sampling_rate:
|
60 |
-
# Pad the audio if it's shorter than the desired length
|
61 |
-
padding = torch.zeros(1, self.max_audio_len * self.sampling_rate - len_audio)
|
62 |
-
speech_array = torch.cat([speech_array, padding], dim=1)
|
63 |
-
else:
|
64 |
-
# Truncate the audio if it's longer than the desired length
|
65 |
-
speech_array = speech_array[:, :self.max_audio_len * self.sampling_rate]
|
66 |
-
|
67 |
-
speech_array = speech_array.squeeze().numpy()
|
68 |
-
|
69 |
-
return {"input_values": speech_array, "attention_mask": None}
|
70 |
-
|
71 |
-
|
72 |
-
class CollateFunc:
|
73 |
-
def __init__(
|
74 |
-
self,
|
75 |
-
processor: Wav2Vec2Processor,
|
76 |
-
padding: Union[bool, str] = True,
|
77 |
-
pad_to_multiple_of: Optional[int] = None,
|
78 |
-
return_attention_mask: bool = True,
|
79 |
-
sampling_rate: int = 16000,
|
80 |
-
max_length: Optional[int] = None,
|
81 |
-
):
|
82 |
-
self.sampling_rate = sampling_rate
|
83 |
-
self.processor = processor
|
84 |
-
self.padding = padding
|
85 |
-
self.pad_to_multiple_of = pad_to_multiple_of
|
86 |
-
self.return_attention_mask = return_attention_mask
|
87 |
-
self.max_length = max_length
|
88 |
-
|
89 |
-
def __call__(self, batch: List[Dict[str, np.ndarray]]):
|
90 |
-
# Extract input_values from the batch
|
91 |
-
input_values = [item["input_values"] for item in batch]
|
92 |
-
|
93 |
-
batch = self.processor(
|
94 |
-
input_values,
|
95 |
-
sampling_rate=self.sampling_rate,
|
96 |
-
return_tensors="pt",
|
97 |
-
padding=self.padding,
|
98 |
-
max_length=self.max_length,
|
99 |
-
pad_to_multiple_of=self.pad_to_multiple_of,
|
100 |
-
return_attention_mask=self.return_attention_mask
|
101 |
-
)
|
102 |
-
|
103 |
-
return {
|
104 |
-
"input_values": batch.input_values,
|
105 |
-
"attention_mask": batch.attention_mask if self.return_attention_mask else None
|
106 |
-
}
|
107 |
-
|
108 |
-
|
109 |
-
def predict(test_dataloader, model, device: torch.device):
|
110 |
-
"""
|
111 |
-
Predict the class of the audio
|
112 |
-
"""
|
113 |
-
model.to(device)
|
114 |
-
model.eval()
|
115 |
-
preds = []
|
116 |
-
|
117 |
-
with torch.no_grad():
|
118 |
-
for batch in tqdm.tqdm(test_dataloader):
|
119 |
-
input_values, attention_mask = batch['input_values'].to(device), batch['attention_mask'].to(device)
|
120 |
-
|
121 |
-
logits = model(input_values, attention_mask=attention_mask).logits
|
122 |
-
scores = F.softmax(logits, dim=-1)
|
123 |
-
|
124 |
-
pred = torch.argmax(scores, dim=1).cpu().detach().numpy()
|
125 |
-
|
126 |
-
preds.extend(pred)
|
127 |
-
|
128 |
-
return preds
|
129 |
-
|
130 |
-
|
131 |
-
def get_gender(model_name_or_path: str, audio_paths: List[str], label2id: Dict, id2label: Dict, device: torch.device):
|
132 |
-
num_labels = 2
|
133 |
-
|
134 |
-
feature_extractor = AutoFeatureExtractor.from_pretrained(model_name_or_path)
|
135 |
-
model = AutoModelForAudioClassification.from_pretrained(
|
136 |
-
pretrained_model_name_or_path=model_name_or_path,
|
137 |
-
num_labels=num_labels,
|
138 |
-
label2id=label2id,
|
139 |
-
id2label=id2label,
|
140 |
-
)
|
141 |
-
|
142 |
-
test_dataset = CustomDataset(audio_paths, max_audio_len=5) # for 5-second audio
|
143 |
-
|
144 |
-
data_collator = CollateFunc(
|
145 |
-
processor=feature_extractor,
|
146 |
-
padding=True,
|
147 |
-
sampling_rate=16000,
|
148 |
-
)
|
149 |
-
|
150 |
-
test_dataloader = DataLoader(
|
151 |
-
dataset=test_dataset,
|
152 |
-
batch_size=16,
|
153 |
-
collate_fn=data_collator,
|
154 |
-
shuffle=False,
|
155 |
-
num_workers=2
|
156 |
-
)
|
157 |
-
|
158 |
-
preds = predict(test_dataloader=test_dataloader, model=model, device=device)
|
159 |
-
|
160 |
-
return preds
|
161 |
-
|
162 |
-
model_name_or_path = "alefiury/wav2vec2-large-xlsr-53-gender-recognition-librispeech"
|
163 |
-
|
164 |
-
audio_paths = [] # Must be a list with absolute paths of the audios that will be used in inference
|
165 |
-
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
|
166 |
|
167 |
-
label2id = {
|
168 |
-
"female": 0,
|
169 |
-
"male": 1
|
170 |
-
}
|
171 |
|
172 |
-
|
173 |
-
|
174 |
-
|
175 |
-
|
|
|
|
|
|
|
176 |
|
177 |
-
num_labels = 2
|
178 |
|
179 |
-
|
|
|
|
|
|
1 |
+
import re
|
2 |
+
from gender_prediction import get_gender
|
3 |
+
import gradio as gr
|
|
|
|
|
|
|
4 |
import torch
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
5 |
|
|
|
|
|
|
|
|
|
6 |
|
7 |
+
def app(voice):
|
8 |
+
model_name_or_path = "alefiury/wav2vec2-large-xlsr-53-gender-recognition-librispeech"
|
9 |
+
audio_paths = [voice]
|
10 |
+
device = torch.device("cuda" if torch.cuda.is_available() else "cpu")
|
11 |
+
predicted_label = get_gender(model_name_or_path, audio_paths, device)
|
12 |
+
gender=re.search("female|male",predicted_label)
|
13 |
+
return gender.string
|
14 |
|
|
|
15 |
|
16 |
+
interface=gr.Interface(fn=app,inputs=[gr.components.Audio(type="filepath",sources="upload",label="upload voice")],
|
17 |
+
outputs=[gr.components.Textbox(label="your result")])
|
18 |
+
interface.launch(debug=True)
|