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| import os | |
| from huggingface_hub import hf_hub_download | |
| import gradio as gr | |
| import json | |
| import pandas as pd | |
| import collections | |
| import scipy.signal | |
| import numpy as np | |
| from functools import partial | |
| from openwakeword.model import Model | |
| from openwakeword.utils import download_models | |
| download_models() | |
| # 用 Secret token 從 HF Model Hub 下載私有模型 | |
| hf_token = os.environ.get("HF_TOKEN") | |
| model_path = hf_hub_download( | |
| repo_id="JTBTechnology/kmu_wakeword", | |
| filename="hi_kmu_0721.onnx", # 改成你模型內的正確檔名 | |
| token=hf_token, | |
| repo_type="model" | |
| ) | |
| # 直接用下載的模型路徑載入 | |
| model = Model(wakeword_models=[model_path], inference_framework="onnx") | |
| # Define function to process audio | |
| # def process_audio(audio, state=collections.defaultdict(partial(collections.deque, maxlen=60))): | |
| def process_audio(audio, state=None): | |
| if state is None: | |
| state = collections.defaultdict(partial(collections.deque, maxlen=60)) | |
| # Resample audio to 16khz if needed | |
| if audio[0] != 16000: | |
| data = scipy.signal.resample(audio[1], int(float(audio[1].shape[0])/audio[0]*16000)) | |
| # Get predictions | |
| for i in range(0, data.shape[0], 1280): | |
| if len(data.shape) == 2 or data.shape[-1] == 2: | |
| chunk = data[i:i+1280][:, 0] # just get one channel of audio | |
| else: | |
| chunk = data[i:i+1280] | |
| if chunk.shape[0] == 1280: | |
| prediction = model.predict(chunk) | |
| for key in prediction: | |
| #Fill deque with zeros if it's empty | |
| if len(state[key]) == 0: | |
| state[key].extend(np.zeros(60)) | |
| # Add prediction | |
| state[key].append(prediction[key]) | |
| # Make line plot | |
| dfs = [] | |
| for key in state.keys(): | |
| df = pd.DataFrame({"x": np.arange(len(state[key])), "y": state[key], "Model": key}) | |
| dfs.append(df) | |
| df = pd.concat(dfs) | |
| plot = gr.LinePlot( | |
| value=df, | |
| x='x', | |
| y='y', | |
| color="Model", | |
| y_lim=(0,1), | |
| tooltip="Model", | |
| width=600, | |
| height=300, | |
| x_title="Time (frames)", | |
| y_title="Model Score", | |
| color_legend_position="bottom" | |
| ) | |
| # 1. 將 state 轉成可 JSON 序列化格式(dict of lists) | |
| serializable_state = {k: [float(x) for x in v] for k, v in state.items()} | |
| # 2. 回傳 serializable_state 給 Gradio | |
| return plot, serializable_state | |
| # Create Gradio interface and launch | |
| desc = """ | |
| 這是 [openWakeWord](https://github.com/dscripka/openWakeWord) 最新版本預設模型的小工具示範。 | |
| 請點一下下面的「開始錄音」按鈕,就能直接用麥克風測試。 | |
| 系統會即時把每個模型的分數用折線圖秀出來,你也可以把滑鼠移到線上看是哪一個模型。 | |
| 每一個模型都有自己專屬的喚醒詞或指令句(更多可以參考 [模型說明](https://github.com/dscripka/openWakeWord/tree/main/docs/models))。 | |
| 如果偵測到你講了對的關鍵詞,圖上對應模型的分數會突然變高。你可以試著講下面的範例語句試試看: | |
| | 模型名稱 | 建議語句 | | |
| | ------------- | ------ | | |
| | hi\_kmu\_0721 | 「嗨,高醫」 | | |
| """ | |
| gr_int = gr.Interface( | |
| title = "語音喚醒展示", | |
| description = desc, | |
| css = ".flex {flex-direction: column} .gr-panel {width: 100%}", | |
| fn=process_audio, | |
| inputs=[ | |
| gr.Audio(sources=["microphone"], type="numpy", streaming=True, show_label=False), | |
| "state" | |
| ], | |
| outputs=[ | |
| gr.LinePlot(show_label=False), | |
| "state" | |
| ], | |
| live=True) | |
| gr_int.launch() |