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Browse files- .gitignore +1 -0
- README.md +6 -5
- app.py +110 -0
- packages.txt +1 -0
- requirements.txt +2 -0
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.idea
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README.md
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---
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title: Kotoba Whisper
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sdk: gradio
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sdk_version:
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app_file: app.py
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pinned: false
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---
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Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
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---
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title: Kotoba Whisper Demo
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emoji: 🔥
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colorFrom: yellow
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colorTo: blue
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sdk: gradio
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sdk_version: 4.39.0
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app_file: app.py
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pinned: false
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license: apache-2.0
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---
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Check out the configuration reference at https://huggingface.co/docs/hub/spaces-config-reference
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app.py
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import os
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from math import floor
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from typing import Optional
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import spaces
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import torch
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import gradio as gr
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from transformers import pipeline
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from transformers.pipelines.audio_utils import ffmpeg_read
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# configuration
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MODEL_NAME = "japanese-asr/distil-whisper-bilingual-v1.0"
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BATCH_SIZE = 16
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CHUNK_LENGTH_S = 15
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# device setting
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if torch.cuda.is_available():
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torch_dtype = torch.bfloat16
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device = "cuda"
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model_kwargs = {'attn_implementation': 'sdpa'}
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else:
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torch_dtype = torch.float32
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device = "cpu"
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model_kwargs = {}
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# define the pipeline
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pipe = pipeline(
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model=MODEL_NAME,
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chunk_length_s=CHUNK_LENGTH_S,
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batch_size=BATCH_SIZE,
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torch_dtype=torch_dtype,
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device=device,
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model_kwargs=model_kwargs,
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trust_remote_code=True
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)
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def format_time(start: Optional[float], end: Optional[float]):
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def _format_time(seconds: Optional[float]):
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if seconds is None:
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return "complete "
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minutes = floor(seconds / 60)
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hours = floor(seconds / 3600)
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seconds = seconds - hours * 3600 - minutes * 60
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m_seconds = floor(round(seconds - floor(seconds), 3) * 10 ** 3)
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seconds = floor(seconds)
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return f'{hours:02}:{minutes:02}:{seconds:02}.{m_seconds:03}'
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return f"[{_format_time(start)}-> {_format_time(end)}]:"
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@spaces.GPU
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def get_prediction(inputs, task: str, language: Optional[str]):
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generate_kwargs = {"task": task}
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if language:
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generate_kwargs['language'] = language
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prediction = pipe(inputs, return_timestamps=True, generate_kwargs=generate_kwargs)
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text = "".join([c['text'] for c in prediction['chunks']])
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text_timestamped = "\n".join([
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f"{format_time(*c['timestamp'])} {c['text']}" for c in prediction['chunks']
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])
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return text, text_timestamped
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def transcribe(inputs: str, task: str, language: str):
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language = None if language == "none" else language
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if inputs is None:
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raise gr.Error("No audio file submitted! Please upload or record an audio file before submitting your request.")
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with open(inputs, "rb") as f:
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inputs = f.read()
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inputs = ffmpeg_read(inputs, pipe.feature_extractor.sampling_rate)
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inputs = {"array": inputs, "sampling_rate": pipe.feature_extractor.sampling_rate}
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return get_prediction(inputs, task, language)
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demo = gr.Blocks()
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description = (f"Transcribe long-form microphone or audio inputs with the click of a button! Demo uses Kotoba-Whisper "
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f"checkpoint [{MODEL_NAME}](https://huggingface.co/{MODEL_NAME}) and 🤗 Transformers to transcribe audio"
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f" files of arbitrary length.")
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title = f"Transcribe Audio with {os.path.basename(MODEL_NAME)}"
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mf_transcribe = gr.Interface(
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fn=transcribe,
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inputs=[
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gr.Audio(sources="microphone", type="filepath"),
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gr.Textbox(lines=1, placeholder="Prompt"),
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gr.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
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gr.Radio(["none", "ja", "en"], label="Language", default="none")
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],
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outputs=["text", "text"],
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title=title,
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description=description,
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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", type="filepath", label="Audio file"),
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gr.Textbox(lines=1, placeholder="Prompt"),
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gr.Radio(["transcribe", "translate"], label="Task", default="transcribe"),
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gr.Radio(["none", "ja", "en"], label="Language", default="none")
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],
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outputs=["text", "text"],
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title=title,
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description=description,
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allow_flagging="never",
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)
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with demo:
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gr.TabbedInterface([mf_transcribe, file_transcribe], ["Microphone", "Audio file"])
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demo.queue(api_open=False, default_concurrency_limit=40).launch(show_api=False, show_error=True)
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packages.txt
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@@ -0,0 +1 @@
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ffmpeg
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requirements.txt
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git+https://github.com/huggingface/transformers
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torch
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