Spaces:
Running
Running
Update app.py
Browse files
app.py
CHANGED
@@ -10,16 +10,13 @@ import time
|
|
10 |
import os
|
11 |
import numpy as np
|
12 |
|
13 |
-
# Constants
|
14 |
MODEL_NAME = "dataprizma/whisper-large-v3-turbo"
|
15 |
BATCH_SIZE = 8
|
16 |
FILE_LIMIT_MB = 1000
|
17 |
YT_LENGTH_LIMIT_S = 3600 # 1 hour limit
|
18 |
|
19 |
-
# Device selection
|
20 |
device = 0 if torch.cuda.is_available() else "cpu"
|
21 |
|
22 |
-
# Load Whisper pipeline
|
23 |
pipe = pipeline(
|
24 |
task="automatic-speech-recognition",
|
25 |
model=MODEL_NAME,
|
@@ -31,35 +28,29 @@ pipe = pipeline(
|
|
31 |
},
|
32 |
)
|
33 |
|
34 |
-
|
35 |
-
def transcribe(audio_file, task):
|
36 |
if audio_file is None:
|
37 |
raise gr.Error("No audio file submitted! Please upload or record an audio file before submitting.")
|
38 |
|
39 |
-
# Open file as binary to ensure correct data type
|
40 |
with open(audio_file, "rb") as f:
|
41 |
audio_data = f.read()
|
42 |
|
43 |
-
# Read audio using ffmpeg_read (correcting input format)
|
44 |
audio_array = ffmpeg_read(audio_data, pipe.feature_extractor.sampling_rate)
|
45 |
-
|
46 |
duration = len(audio_array) / pipe.feature_extractor.sampling_rate
|
47 |
print(f"Audio duration: {duration:.2f} seconds")
|
48 |
|
49 |
-
# Convert to proper format
|
50 |
inputs = {
|
51 |
"array": np.array(audio_array),
|
52 |
"sampling_rate": pipe.feature_extractor.sampling_rate
|
53 |
}
|
54 |
|
55 |
generate_kwargs = {
|
56 |
-
"task":
|
57 |
-
"no_speech_threshold": 0.
|
58 |
"logprob_threshold": -1.0,
|
59 |
"compression_ratio_threshold": 2.4
|
60 |
}
|
61 |
-
|
62 |
-
# Perform transcription
|
63 |
result = pipe(
|
64 |
inputs,
|
65 |
batch_size=BATCH_SIZE,
|
@@ -69,19 +60,16 @@ def transcribe(audio_file, task):
|
|
69 |
|
70 |
return result["text"]
|
71 |
|
72 |
-
# Gradio UI
|
73 |
demo = gr.Blocks()
|
74 |
|
75 |
file_transcribe = gr.Interface(
|
76 |
fn=transcribe,
|
77 |
inputs=[
|
78 |
gr.Audio(type="filepath", label="Audio file"),
|
79 |
-
gr.Radio(["transcribe", "translate"], label="Task"),
|
80 |
],
|
81 |
outputs="text",
|
82 |
title="Whisper Large V3: Transcribe Audio",
|
83 |
description="Whisper Large V3 fine-tuned for Uzbek language by Dataprizma",
|
84 |
-
flagging_mode="never",
|
85 |
)
|
86 |
|
87 |
with demo:
|
|
|
10 |
import os
|
11 |
import numpy as np
|
12 |
|
|
|
13 |
MODEL_NAME = "dataprizma/whisper-large-v3-turbo"
|
14 |
BATCH_SIZE = 8
|
15 |
FILE_LIMIT_MB = 1000
|
16 |
YT_LENGTH_LIMIT_S = 3600 # 1 hour limit
|
17 |
|
|
|
18 |
device = 0 if torch.cuda.is_available() else "cpu"
|
19 |
|
|
|
20 |
pipe = pipeline(
|
21 |
task="automatic-speech-recognition",
|
22 |
model=MODEL_NAME,
|
|
|
28 |
},
|
29 |
)
|
30 |
|
31 |
+
def transcribe(audio_file):
|
|
|
32 |
if audio_file is None:
|
33 |
raise gr.Error("No audio file submitted! Please upload or record an audio file before submitting.")
|
34 |
|
|
|
35 |
with open(audio_file, "rb") as f:
|
36 |
audio_data = f.read()
|
37 |
|
|
|
38 |
audio_array = ffmpeg_read(audio_data, pipe.feature_extractor.sampling_rate)
|
|
|
39 |
duration = len(audio_array) / pipe.feature_extractor.sampling_rate
|
40 |
print(f"Audio duration: {duration:.2f} seconds")
|
41 |
|
|
|
42 |
inputs = {
|
43 |
"array": np.array(audio_array),
|
44 |
"sampling_rate": pipe.feature_extractor.sampling_rate
|
45 |
}
|
46 |
|
47 |
generate_kwargs = {
|
48 |
+
"task": "transcribe",
|
49 |
+
"no_speech_threshold": 0.4,
|
50 |
"logprob_threshold": -1.0,
|
51 |
"compression_ratio_threshold": 2.4
|
52 |
}
|
53 |
+
|
|
|
54 |
result = pipe(
|
55 |
inputs,
|
56 |
batch_size=BATCH_SIZE,
|
|
|
60 |
|
61 |
return result["text"]
|
62 |
|
|
|
63 |
demo = gr.Blocks()
|
64 |
|
65 |
file_transcribe = gr.Interface(
|
66 |
fn=transcribe,
|
67 |
inputs=[
|
68 |
gr.Audio(type="filepath", label="Audio file"),
|
|
|
69 |
],
|
70 |
outputs="text",
|
71 |
title="Whisper Large V3: Transcribe Audio",
|
72 |
description="Whisper Large V3 fine-tuned for Uzbek language by Dataprizma",
|
|
|
73 |
)
|
74 |
|
75 |
with demo:
|