Create app.py
Browse files
app.py
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import os
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import time
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import warnings
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from pathlib import Path
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import gradio as gr
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import librosa
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import spaces
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import torch
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from loguru import logger
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from transformers import pipeline
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warnings.filterwarnings("ignore")
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is_hf = os.getenv("SYSTEM") == "spaces"
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generate_kwargs = {
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"language": "Japanese",
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"do_sample": False,
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"num_beams": 1,
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"no_repeat_ngram_size": 5,
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"max_new_tokens": 64,
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}
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model_dict = {
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"whisper-large-v3-turbo": "openai/whisper-large-v3-turbo",
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"kotoba-whisper-v2.0": "kotoba-tech/kotoba-whisper-v2.0",
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"anime-whisper": "litagin/anime-whisper",
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}
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logger.info("Initializing pipelines...")
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pipe_dict = {
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k: pipeline(
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"automatic-speech-recognition",
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model=v,
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device="cuda" if torch.cuda.is_available() else "cpu",
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torch_dtype=torch.float16 if torch.cuda.is_available() else torch.float32,
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)
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for k, v in model_dict.items()
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}
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logger.success("Pipelines initialized!")
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@spaces.GPU
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def transcribe_common(audio: str, model: str) -> str:
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if not audio:
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return "No audio file"
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filename = Path(audio).name
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logger.info(f"Model: {model}")
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logger.info(f"Audio: {filename}")
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# Read and resample audio to 16kHz
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try:
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y, sr = librosa.load(audio, mono=True, sr=16000)
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except Exception as e:
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# First convert to wav if librosa cannot read the file
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logger.error(f"Error reading file: {e}")
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from pydub import AudioSegment
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audio = AudioSegment.from_file(audio)
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audio.export("temp.wav", format="wav")
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y, sr = librosa.load("temp.wav", mono=True, sr=16000)
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Path("temp.wav").unlink()
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# Get duration of audio
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duration = librosa.get_duration(y=y, sr=sr)
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logger.info(f"Duration: {duration:.2f}s")
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if duration > 15:
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logger.error(f"Audio too long, limit is 15 seconds, got {duration:.2f}s")
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return f"Audio too long, limit is 15 seconds, got {duration:.2f}s"
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start_time = time.time()
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result = pipe_dict[model](y, generate_kwargs=generate_kwargs)["text"]
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end_time = time.time()
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logger.success(f"Finished in {end_time - start_time:.2f}s\n{result}")
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return result
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def transcribe_others(audio) -> tuple[str, str]:
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result_v3 = transcribe_common(audio, "whisper-large-v3-turbo")
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result_kotoba_v2 = transcribe_common(audio, "kotoba-whisper-v2.0")
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return result_v3, result_kotoba_v2
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def transcribe_anime_whisper(audio) -> str:
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return transcribe_common(audio, "anime-whisper")
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initial_md = """
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# Anime-Whisper Demo
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[**Anime Whisper**](https://huggingface.co/litagin/anime-whisper): 5千時間以上のアニメ調セリフと台本でファインチューニングされた日本語音声認識モデルのデモです。句読点や感嘆符がリズムや感情に合わせて自然に付き、NSFW含む非言語発話もうまく台本調に書き起こされます。
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- デモでは**音声は15秒まで**しか受け付けません
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- 日本語のみ対応 (Japanese only)
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- 比較のために [openai/whisper-large-v3-turbo](https://huggingface.co/openai/whisper-large-v3-turbo) と [kotoba-tech/kotoba-whisper-v2.0](https://huggingface.co/kotoba-tech/kotoba-whisper-v2.0) も用意しています
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pipeに渡しているkwargsは以下:
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```python
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generate_kwargs = {
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"language": "Japanese",
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"do_sample": False,
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"num_beams": 1,
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"no_repeat_ngram_size": 5,
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"max_new_tokens": 64, # 結果が長いときは途中で打ち切られる
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}
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```
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"""
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with gr.Blocks() as app:
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gr.Markdown(initial_md)
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audio = gr.Audio(type="filepath")
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with gr.Row():
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with gr.Column():
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gr.Markdown("### Anime-Whisper")
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button_galgame = gr.Button("Transcribe with Anime-Whisper")
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output_galgame = gr.Textbox(label="Result")
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gr.Markdown("### Comparison")
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button_others = gr.Button("Transcribe with other models")
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with gr.Row():
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with gr.Column():
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gr.Markdown("### Whisper-Large-V3-Turbo")
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output_v3 = gr.Textbox(label="Result")
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with gr.Column():
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gr.Markdown("### Kotoba-Whisper-V2.0")
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output_kotoba_v2 = gr.Textbox(label="Result")
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button_galgame.click(
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transcribe_anime_whisper,
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inputs=[audio],
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outputs=[output_galgame],
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)
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button_others.click(
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transcribe_others,
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inputs=[audio],
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outputs=[output_v3, output_kotoba_v2],
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)
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app.launch(inbrowser=True)
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