Spaces:
Running
Running
Commit
Β·
2647bd6
1
Parent(s):
895c600
fix: enhance model loading and selection in transcribe function for improved user experience
Browse files
app.py
CHANGED
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@@ -10,28 +10,42 @@ import librosa
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# Important: Don't initialize CUDA in the main process for Spaces
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# The model will be loaded in the worker process through the GPU decorator
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model = None
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# This function will be called in the GPU worker process
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global model
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print(f"CUDA available: {torch.cuda.is_available()}")
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if torch.cuda.is_available():
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print(f"CUDA device: {torch.cuda.get_device_name(0)}")
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print(f"Model loaded on device: {model.device}")
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return model
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@spaces.GPU(duration=120)
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def transcribe(audio, state="", audio_buffer=None, last_processed_time=0):
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# Load the model inside the GPU worker process
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import numpy as np
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import soundfile as sf
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import librosa
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import os
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model = load_model()
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if audio_buffer is None:
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audio_buffer = []
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@@ -129,7 +143,22 @@ def transcribe(audio, state="", audio_buffer=None, last_processed_time=0):
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# Define the Gradio interface
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with gr.Blocks(title="Real-time Speech-to-Text with NeMo") as demo:
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gr.Markdown("# ποΈ Real-time Speech-to-Text Transcription")
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gr.Markdown("Powered by NVIDIA NeMo
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with gr.Row():
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with gr.Column(scale=2):
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audio_input = gr.Audio(
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@@ -152,19 +181,30 @@ with gr.Blocks(title="Real-time Speech-to-Text with NeMo") as demo:
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placeholder="Real-time results will appear here...",
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lines=2
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)
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# State to store the ongoing transcription
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state = gr.State("")
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audio_buffer = gr.State(value=None)
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last_processed_time = gr.State(value=0)
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# Handle the audio stream
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audio_input.stream(
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fn=transcribe,
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inputs=[audio_input, state, audio_buffer, last_processed_time],
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outputs=[state, streaming_text, audio_buffer, last_processed_time],
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)
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# Clear the transcription
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def clear_transcription():
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return "", "", None, 0
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@@ -180,13 +220,14 @@ with gr.Blocks(title="Real-time Speech-to-Text with NeMo") as demo:
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inputs=[state],
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outputs=[text_output]
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)
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gr.Markdown("## π Instructions")
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gr.Markdown("""
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1.
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2.
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3.
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4.
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""")
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# Launch the app
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# Important: Don't initialize CUDA in the main process for Spaces
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# The model will be loaded in the worker process through the GPU decorator
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model = None
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current_model_name = "nvidia/parakeet-tdt-0.6b-v2"
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# Available models
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available_models = ["nvidia/parakeet-tdt-0.6b-v2"]
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def load_model(model_name=None):
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# This function will be called in the GPU worker process
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global model, current_model_name
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# Use the specified model name or the current one
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model_name = model_name or current_model_name
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# Check if we need to load a new model
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if model is None or model_name != current_model_name:
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print(f"Loading model {model_name} in worker process")
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print(f"CUDA available: {torch.cuda.is_available()}")
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if torch.cuda.is_available():
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print(f"CUDA device: {torch.cuda.get_device_name(0)}")
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# Update the current model name
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current_model_name = model_name
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# Load the selected model
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model = nemo_asr.models.EncDecRNNTBPEModel.from_pretrained(model_name)
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print(f"Model loaded on device: {model.device}")
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return model
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@spaces.GPU(duration=120)
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def transcribe(audio, model_name="nvidia/parakeet-tdt-0.6b-v2", state="", audio_buffer=None, last_processed_time=0):
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# Load the model inside the GPU worker process
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import numpy as np
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import soundfile as sf
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import librosa
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import os
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model = load_model(model_name)
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if audio_buffer is None:
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audio_buffer = []
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# Define the Gradio interface
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with gr.Blocks(title="Real-time Speech-to-Text with NeMo") as demo:
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gr.Markdown("# ποΈ Real-time Speech-to-Text Transcription")
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gr.Markdown("Powered by NVIDIA NeMo")
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# Model selection and loading
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with gr.Row():
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with gr.Column(scale=3):
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model_dropdown = gr.Dropdown(
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choices=available_models,
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value=current_model_name,
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label="Select ASR Model"
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)
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with gr.Column(scale=1):
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load_button = gr.Button("Load Selected Model")
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# Status indicator for model loading
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model_status = gr.Textbox(value=f"Current model: {current_model_name}", label="Model Status")
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with gr.Row():
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with gr.Column(scale=2):
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audio_input = gr.Audio(
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placeholder="Real-time results will appear here...",
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lines=2
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)
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# State to store the ongoing transcription
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state = gr.State("")
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audio_buffer = gr.State(value=None)
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last_processed_time = gr.State(value=0)
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# Function to handle model selection
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def update_model(model_name):
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global current_model_name
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current_model_name = model_name
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return f"Current model: {model_name}", None, 0 # Reset audio buffer and last processed time
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# Load model button event
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load_button.click(
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fn=update_model,
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inputs=[model_dropdown],
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outputs=[model_status, audio_buffer, last_processed_time]
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)
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# Handle the audio stream
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audio_input.stream(
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fn=transcribe,
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inputs=[audio_input, model_dropdown, state, audio_buffer, last_processed_time],
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outputs=[state, streaming_text, audio_buffer, last_processed_time],
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) # Clear the transcription
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def clear_transcription():
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return "", "", None, 0
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inputs=[state],
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outputs=[text_output]
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)
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gr.Markdown("## π Instructions")
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gr.Markdown("""
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1. Select an ASR model from the dropdown menu
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2. Click 'Load Selected Model' to load the model
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3. Click the microphone button to start recording
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4. Speak clearly into your microphone
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5. The transcription will appear in real-time
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6. Click 'Clear Transcript' to start a new transcription
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""")
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# Launch the app
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