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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() |