File size: 2,464 Bytes
ed97bcc
 
 
 
966d76f
ed97bcc
 
1500574
ed97bcc
 
 
 
1500574
ed97bcc
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1500574
ed97bcc
1500574
ed97bcc
 
 
 
 
 
 
 
1500574
ed97bcc
 
1500574
966d76f
ed97bcc
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1500574
966d76f
 
1500574
8472c6f
1500574
966d76f
 
1500574
 
966d76f
 
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 torch
# import torchaudio
# import gradio as gr
# from transformers import Wav2Vec2BertProcessor, Wav2Vec2BertForCTC

# # Set device
# device = "cuda" if torch.cuda.is_available() else "cpu"

# # Load processor & model
# model_name = "cdactvm/w2v-bert-punjabi"  # Change if using a Punjabi ASR model
# processor = Wav2Vec2BertProcessor.from_pretrained(model_name)
# model = Wav2Vec2BertForCTC.from_pretrained(model_name, torch_dtype=torch.bfloat16).to(device)

# def transcribe(audio_path):
#     # Load audio file
#     waveform, sample_rate = torchaudio.load(audio_path)

#     # Convert stereo to mono (if needed)
#     if waveform.shape[0] > 1:
#         waveform = torch.mean(waveform, dim=0, keepdim=True)

#     # Resample to 16kHz
#     if sample_rate != 16000:
#         waveform = torchaudio.transforms.Resample(orig_freq=sample_rate, new_freq=16000)(waveform)

#     # Process audio
#     inputs = processor(waveform.squeeze(0), sampling_rate=16000, return_tensors="pt")
#     inputs = {key: val.to(device, dtype=torch.bfloat16) for key, val in inputs.items()}

#     # Get logits & transcribe
#     with torch.no_grad():
#         logits = model(**inputs).logits
#     predicted_ids = torch.argmax(logits, dim=-1)
#     transcription = processor.batch_decode(predicted_ids)[0]

#     return transcription

# # Gradio Interface
# app = gr.Interface(
#     fn=transcribe,
#     inputs=gr.Audio(sources="upload", type="filepath"),
#     outputs="text",
#     title="Punjabi Speech-to-Text",
#     description="Upload an audio file and get the transcription in Punjabi."
# )

# if __name__ == "__main__":
#     app.launch()


import gradio as gr
import torch
from transformers import pipeline

# Set device
device = "cuda" if torch.cuda.is_available() else "cpu"

# Load ASR pipeline
asr_pipeline = pipeline(
    "automatic-speech-recognition",
    model="cdactvm/w2v-bert-punjabi",  # Replace with a Punjabi ASR model if available
    torch_dtype=torch.bfloat16,
    device=0 if torch.cuda.is_available() else -1  # GPU (0) or CPU (-1)
)

def transcribe(audio_path):
    # Run inference
    result = asr_pipeline(audio_path)
    return result["text"]

# Gradio Interface
app = gr.Interface(
    fn=transcribe,
    inputs=gr.Audio(sources="upload", type="filepath"),
    outputs="text",
    title="Punjabi Speech-to-Text",
    description="Upload an audio file and get the transcription in Punjabi."
)

if __name__ == "__main__":
    app.launch()