Spaces:
Sleeping
Sleeping
Added ASR model files
Browse files- README.md +14 -0
- app.py +107 -0
- requirements.txt +5 -0
README.md
CHANGED
@@ -10,3 +10,17 @@ pinned: false
|
|
10 |
---
|
11 |
|
12 |
Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
10 |
---
|
11 |
|
12 |
Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
|
13 |
+
|
14 |
+
# ASR Transcription Service
|
15 |
+
|
16 |
+
This is an Automatic Speech Recognition (ASR) system deployed on Hugging Face Spaces using Gradio.
|
17 |
+
|
18 |
+
## Features
|
19 |
+
- Supports **Tunisian Arabic (tn)**, **French (fr)**, and **English (en)**
|
20 |
+
- Voice Activity Detection (VAD) for noise removal
|
21 |
+
- Model based on OpenAI Whisper and a fine-tuned Tunisian ASR model
|
22 |
+
|
23 |
+
## How to Use
|
24 |
+
1. Upload an audio file or record using the microphone.
|
25 |
+
2. Select the transcription language.
|
26 |
+
3. Get the transcribed text!
|
app.py
ADDED
@@ -0,0 +1,107 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
1 |
+
import gradio as gr
|
2 |
+
import torch
|
3 |
+
import librosa
|
4 |
+
import numpy as np
|
5 |
+
import webrtcvad
|
6 |
+
from transformers import WhisperProcessor, WhisperForConditionalGeneration, pipeline
|
7 |
+
|
8 |
+
# Model names
|
9 |
+
TN_MODEL_NAME = "amenIKh/Tunisian_Checkpoint12"
|
10 |
+
WHISPER_MODEL_NAME = "openai/whisper-small"
|
11 |
+
|
12 |
+
# Initialize pipelines
|
13 |
+
pipe_tn = pipeline(
|
14 |
+
task="automatic-speech-recognition",
|
15 |
+
model=TN_MODEL_NAME,
|
16 |
+
device=0 if torch.cuda.is_available() else -1,
|
17 |
+
)
|
18 |
+
|
19 |
+
# Load Whisper model and processor
|
20 |
+
whisper_model = WhisperForConditionalGeneration.from_pretrained(WHISPER_MODEL_NAME)
|
21 |
+
whisper_processor = WhisperProcessor.from_pretrained(WHISPER_MODEL_NAME)
|
22 |
+
|
23 |
+
device = "cuda" if torch.cuda.is_available() else "cpu"
|
24 |
+
whisper_model.to(device)
|
25 |
+
|
26 |
+
# Function to apply VAD
|
27 |
+
def apply_vad(audio, sr, frame_duration_ms=30):
|
28 |
+
vad = webrtcvad.Vad()
|
29 |
+
vad.set_mode(3) # Aggressiveness mode, higher value is more aggressive
|
30 |
+
|
31 |
+
frame_size = int(sr * frame_duration_ms / 1000)
|
32 |
+
offset = 0
|
33 |
+
voiced_frames = []
|
34 |
+
|
35 |
+
while offset + frame_size < len(audio):
|
36 |
+
frame = audio[offset:offset + frame_size].astype(np.int16)
|
37 |
+
is_speech = vad.is_speech(frame.tobytes(), sr)
|
38 |
+
|
39 |
+
if is_speech:
|
40 |
+
voiced_frames.append(frame)
|
41 |
+
|
42 |
+
offset += frame_size
|
43 |
+
|
44 |
+
if len(voiced_frames) == 0:
|
45 |
+
return audio # Return original audio if no voiced frames are detected
|
46 |
+
|
47 |
+
voiced_audio = np.concatenate(voiced_frames)
|
48 |
+
return voiced_audio
|
49 |
+
|
50 |
+
# Function to transcribe audio based on language
|
51 |
+
def transcribe_audio(audio, language):
|
52 |
+
try:
|
53 |
+
# Load audio
|
54 |
+
sr = 16000 # Assuming the audio is in 16kHz; adjust if necessary
|
55 |
+
audio, _ = librosa.load(audio, sr=sr)
|
56 |
+
|
57 |
+
# Apply VAD
|
58 |
+
voiced_audio = apply_vad(audio, sr)
|
59 |
+
|
60 |
+
# Select the correct model based on language
|
61 |
+
if language == "tn":
|
62 |
+
result = pipe_tn(voiced_audio)
|
63 |
+
transcription = result.get("text", "")
|
64 |
+
elif language in ["fr", "en"]:
|
65 |
+
forced_decoder_ids = whisper_processor.get_decoder_prompt_ids(language=language, task="transcribe")
|
66 |
+
input_features = whisper_processor(voiced_audio, return_tensors="pt").input_features.to(device)
|
67 |
+
generated_ids = whisper_model.generate(
|
68 |
+
input_features,
|
69 |
+
forced_decoder_ids=forced_decoder_ids
|
70 |
+
)
|
71 |
+
transcription = whisper_processor.batch_decode(generated_ids, skip_special_tokens=True)[0]
|
72 |
+
else:
|
73 |
+
return "Unsupported language specified"
|
74 |
+
|
75 |
+
return transcription
|
76 |
+
except Exception as e:
|
77 |
+
return f"An unexpected error occurred: {str(e)}"
|
78 |
+
# Define Gradio interface
|
79 |
+
def gradio_interface(audio, language):
|
80 |
+
try:
|
81 |
+
# Extract the file path or microphone input from the Gradio audio input
|
82 |
+
if isinstance(audio, tuple):
|
83 |
+
temp_file_path = audio[0] # For microphone recordings, extract file path from the tuple
|
84 |
+
else:
|
85 |
+
temp_file_path = audio # For uploaded files
|
86 |
+
|
87 |
+
# Perform transcription
|
88 |
+
result = transcribe_audio(temp_file_path, language)
|
89 |
+
|
90 |
+
return result
|
91 |
+
except Exception as e:
|
92 |
+
return f"An error occurred: {str(e)}"
|
93 |
+
|
94 |
+
# Create the Gradio app
|
95 |
+
iface = gr.Interface(
|
96 |
+
fn=gradio_interface,
|
97 |
+
inputs=[
|
98 |
+
gr.Audio(sources=["upload","microphone"],type="filepath", label="Upload Audio"),
|
99 |
+
gr.Dropdown(choices=["tn", "fr", "en"], label="Select Language")
|
100 |
+
],
|
101 |
+
outputs="text",
|
102 |
+
title="ASR Transcription Service",
|
103 |
+
description="Upload an audio file and select the language to transcribe the audio."
|
104 |
+
)
|
105 |
+
|
106 |
+
# Add the custom HTML with background image
|
107 |
+
iface.launch()
|
requirements.txt
ADDED
@@ -0,0 +1,5 @@
|
|
|
|
|
|
|
|
|
|
|
|
|
1 |
+
torch
|
2 |
+
gradio
|
3 |
+
transformers
|
4 |
+
librosa
|
5 |
+
webrtcvad
|