Update app.py
Browse files
app.py
CHANGED
@@ -1,53 +1,54 @@
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import torch
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from transformers import pipeline
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from transformers.pipelines.audio_utils import ffmpeg_read
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import gradio as gr
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BATCH_SIZE = 8
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device = 0 if torch.cuda.is_available() else "cpu"
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pipe = pipeline(
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task="automatic-speech-recognition",
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model=MODEL_NAME,
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chunk_length_s=30,
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device=device,
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)
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def format_timestamp(seconds: float, always_include_hours: bool = False, decimal_marker: str = "."):
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if seconds is not None:
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milliseconds = round(seconds * 1000.0)
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-
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hours = milliseconds // 3_600_000
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milliseconds -= hours * 3_600_000
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minutes = milliseconds // 60_000
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milliseconds -= minutes * 60_000
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seconds = milliseconds // 1_000
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milliseconds -= seconds * 1_000
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hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else ""
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return f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}"
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else:
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return seconds
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outputs = pipe(file, batch_size=BATCH_SIZE, generate_kwargs={"task": task}, return_timestamps=return_timestamps)
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text = outputs["text"]
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if return_timestamps:
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timestamps = outputs["chunks"]
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timestamps = [
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f"[{format_timestamp(chunk['timestamp'][0])} -> {format_timestamp(chunk['timestamp'][1])}] {chunk['text']}"
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for chunk in timestamps
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]
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text = "\n".join(str(feature) for feature in timestamps)
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return text
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demo = gr.Blocks()
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mic_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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@@ -57,44 +58,43 @@ mic_transcribe = gr.Interface(
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],
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outputs="text",
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layout="horizontal",
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title="Transcribe Audio",
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description=(
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"Transcribe long-form microphone
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f" checkpoint [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) and 🤗 Transformers to transcribe audio files"
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" of arbitrary length."
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),
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allow_flagging="never",
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)
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file_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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gr.Audio(sources="upload",
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gr.Radio(["transcribe", "translate"], label="Task"),
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gr.Checkbox(label="Return timestamps"),
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],
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outputs="text",
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layout="horizontal",
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title="Transcribe Audio",
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description=(
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"
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f" checkpoint [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) and 🤗 Transformers to transcribe audio files"
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" of arbitrary length."
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),
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examples=[
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["./example.flac", "transcribe", False],
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["./example.flac", "transcribe", True],
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],
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cache_examples=True,
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allow_flagging="never",
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)
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with demo:
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gr.TabbedInterface(
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[mic_transcribe, file_transcribe],
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["Transcribe Microphone", "Transcribe Audio File"]
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)
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import torch
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from transformers import pipeline
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import gradio as gr
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# Define the model details
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MODEL_NAME = "riteshkr/quantized-whisper-large-v3" # Update with your actual model ID
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BATCH_SIZE = 8
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# Select device based on availability of CUDA (GPU) or fallback to CPU
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device = 0 if torch.cuda.is_available() else "cpu"
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# Load the ASR model pipeline
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pipe = pipeline(
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task="automatic-speech-recognition",
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model=MODEL_NAME,
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chunk_length_s=30,
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device=device,
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)
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# Utility function to format timestamps
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def format_timestamp(seconds: float, always_include_hours: bool = False, decimal_marker: str = "."):
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if seconds is not None:
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milliseconds = round(seconds * 1000.0)
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hours = milliseconds // 3_600_000
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milliseconds -= hours * 3_600_000
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minutes = milliseconds // 60_000
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milliseconds -= minutes * 60_000
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seconds = milliseconds // 1_000
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milliseconds -= seconds * 1_000
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hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else ""
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return f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}"
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else:
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return seconds
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# Transcription function for batch processing
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def transcribe(files, task, return_timestamps):
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transcriptions = []
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for file in files: # Process each file in the batch
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outputs = pipe(file, batch_size=BATCH_SIZE, generate_kwargs={"task": task}, return_timestamps=return_timestamps)
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text = outputs["text"]
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if return_timestamps:
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timestamps = outputs["chunks"]
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formatted_chunks = [
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f"[{format_timestamp(chunk['timestamp'][0])} -> {format_timestamp(chunk['timestamp'][1])}] {chunk['text']}"
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for chunk in timestamps
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]
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text = "\n".join(formatted_chunks)
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transcriptions.append(text)
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return "\n\n".join(transcriptions) # Return all transcriptions combined
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# Define Gradio interface for microphone input
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mic_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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],
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outputs="text",
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layout="horizontal",
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title="Whisper Demo: Transcribe Audio",
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description=(
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f"Transcribe long-form microphone inputs with the {MODEL_NAME} model. Supports transcription and translation."
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),
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allow_flagging="never",
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)
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# Define Gradio interface for file upload
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file_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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gr.Audio(sources="upload", type="filepath", label="Upload Audio File"),
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gr.Radio(["transcribe", "translate"], label="Task"),
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gr.Checkbox(label="Return timestamps"),
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],
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outputs="text",
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layout="horizontal",
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title="Whisper Demo: Transcribe Audio",
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description=(
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f"Upload audio files to transcribe or translate them using the {MODEL_NAME} model."
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),
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allow_flagging="never",
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examples=[
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["./example.flac", "transcribe", False],
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["./example.flac", "transcribe", True],
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],
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)
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# Create the Gradio tabbed interface for switching between modes
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demo = gr.Blocks()
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with demo:
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gr.TabbedInterface(
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[mic_transcribe, file_transcribe],
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["Transcribe Microphone", "Transcribe Audio File"]
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)
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# Launch the app
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if __name__ == "__main__":
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demo.launch(debug=True, enable_queue=True, share=True)
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