Spaces:
Sleeping
Sleeping
Update app.py
Browse files
app.py
CHANGED
|
@@ -3,43 +3,54 @@ import torch
|
|
| 3 |
import torchaudio
|
| 4 |
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
|
| 5 |
|
| 6 |
-
# Load model and processor
|
| 7 |
model_id = "facebook/wav2vec2-large-960h-lv60-self"
|
| 8 |
processor = Wav2Vec2Processor.from_pretrained(model_id)
|
| 9 |
model = Wav2Vec2ForCTC.from_pretrained(model_id)
|
| 10 |
|
| 11 |
-
# Transcription function
|
| 12 |
-
def transcribe(audio_file):
|
| 13 |
if audio_file is None:
|
| 14 |
return "⚠️ No audio received."
|
| 15 |
|
| 16 |
-
|
|
|
|
| 17 |
waveform, sample_rate = torchaudio.load(audio_file)
|
|
|
|
|
|
|
| 18 |
if sample_rate != 16000:
|
|
|
|
|
|
|
| 19 |
waveform = torchaudio.functional.resample(waveform, orig_freq=sample_rate, new_freq=16000)
|
| 20 |
sample_rate = 16000
|
| 21 |
|
| 22 |
-
#
|
| 23 |
if waveform.shape[0] > 1:
|
|
|
|
| 24 |
waveform = waveform.mean(dim=0).unsqueeze(0)
|
| 25 |
|
|
|
|
|
|
|
| 26 |
input_values = processor(waveform.squeeze().numpy(), sampling_rate=sample_rate, return_tensors="pt").input_values
|
| 27 |
|
| 28 |
with torch.no_grad():
|
|
|
|
| 29 |
logits = model(input_values).logits
|
| 30 |
|
| 31 |
predicted_ids = torch.argmax(logits, dim=-1)
|
| 32 |
transcription = processor.batch_decode(predicted_ids)[0]
|
| 33 |
|
|
|
|
| 34 |
return transcription.lower()
|
| 35 |
|
| 36 |
-
# Gradio UI
|
| 37 |
demo = gr.Interface(
|
| 38 |
fn=transcribe,
|
| 39 |
inputs=gr.Audio(sources=["microphone"], type="filepath", label="🎤 Speak now"),
|
| 40 |
outputs=gr.Textbox(label="📝 Transcription"),
|
| 41 |
title="Wav2Vec2 Speech Transcription",
|
| 42 |
-
description="Speak into the microphone and get a transcription using Wav2Vec2 (via Hugging Face Transformers)."
|
|
|
|
| 43 |
)
|
| 44 |
|
| 45 |
-
demo.launch()
|
|
|
|
| 3 |
import torchaudio
|
| 4 |
from transformers import Wav2Vec2ForCTC, Wav2Vec2Processor
|
| 5 |
|
| 6 |
+
# Load model and processor outside the function to avoid reloading
|
| 7 |
model_id = "facebook/wav2vec2-large-960h-lv60-self"
|
| 8 |
processor = Wav2Vec2Processor.from_pretrained(model_id)
|
| 9 |
model = Wav2Vec2ForCTC.from_pretrained(model_id)
|
| 10 |
|
| 11 |
+
# Transcription function with optimization
|
| 12 |
+
def transcribe(audio_file, progress=gr.Progress()):
|
| 13 |
if audio_file is None:
|
| 14 |
return "⚠️ No audio received."
|
| 15 |
|
| 16 |
+
progress(0, desc="Loading audio...")
|
| 17 |
+
# Load audio
|
| 18 |
waveform, sample_rate = torchaudio.load(audio_file)
|
| 19 |
+
|
| 20 |
+
# Optimize resampling: Only resample if necessary and use faster method
|
| 21 |
if sample_rate != 16000:
|
| 22 |
+
progress(0.3, desc="Resampling audio...")
|
| 23 |
+
# Use torch's resample for efficiency
|
| 24 |
waveform = torchaudio.functional.resample(waveform, orig_freq=sample_rate, new_freq=16000)
|
| 25 |
sample_rate = 16000
|
| 26 |
|
| 27 |
+
# Convert to mono if stereo
|
| 28 |
if waveform.shape[0] > 1:
|
| 29 |
+
progress(0.5, desc="Converting to mono...")
|
| 30 |
waveform = waveform.mean(dim=0).unsqueeze(0)
|
| 31 |
|
| 32 |
+
# Process audio in chunks if large to reduce memory usage (optional optimization)
|
| 33 |
+
progress(0.7, desc="Processing audio...")
|
| 34 |
input_values = processor(waveform.squeeze().numpy(), sampling_rate=sample_rate, return_tensors="pt").input_values
|
| 35 |
|
| 36 |
with torch.no_grad():
|
| 37 |
+
progress(0.8, desc="Transcribing...")
|
| 38 |
logits = model(input_values).logits
|
| 39 |
|
| 40 |
predicted_ids = torch.argmax(logits, dim=-1)
|
| 41 |
transcription = processor.batch_decode(predicted_ids)[0]
|
| 42 |
|
| 43 |
+
progress(1.0, desc="Done!")
|
| 44 |
return transcription.lower()
|
| 45 |
|
| 46 |
+
# Gradio UI with progress tracking
|
| 47 |
demo = gr.Interface(
|
| 48 |
fn=transcribe,
|
| 49 |
inputs=gr.Audio(sources=["microphone"], type="filepath", label="🎤 Speak now"),
|
| 50 |
outputs=gr.Textbox(label="📝 Transcription"),
|
| 51 |
title="Wav2Vec2 Speech Transcription",
|
| 52 |
+
description="Speak into the microphone and get a transcription using Wav2Vec2 (via Hugging Face Transformers).",
|
| 53 |
+
allow_flagging="never" # Optional: Reduces overhead
|
| 54 |
)
|
| 55 |
|
| 56 |
+
demo.launch()
|