232 lines
8.2 KiB
Python
232 lines
8.2 KiB
Python
import audioop
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import math
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import time
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import threading
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from abc import abstractmethod
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from ai_module.ali_nls import ALiNls
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from ai_module.funasr import FunASR
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from core import wsa_server
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from scheduler.thread_manager import MyThread
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from utils import util
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from utils import config_util as cfg
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import numpy as np
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# 启动时间 (秒)
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_ATTACK = 0.2
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# 释放时间 (秒)
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_RELEASE = 0.75
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class Recorder:
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def __init__(self, fay):
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self.__fay = fay
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self.__running = True
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self.__processing = False
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self.__history_level = []
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self.__history_data = []
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self.__dynamic_threshold = 0.5 # 声音识别的音量阈值
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self.__MAX_LEVEL = 25000
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self.__MAX_BLOCK = 100
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#Edit by xszyou in 20230516:增加本地asr
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self.ASRMode = cfg.ASR_mode
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self.__aLiNls = self.asrclient()
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self.is_awake = False
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self.wakeup_matched = False
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if cfg.config['source']['wake_word_enabled']:
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self.timer = threading.Timer(60, self.reset_wakeup_status) # 60秒后执行reset_wakeup_status方法
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def asrclient(self):
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if self.ASRMode == "ali":
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asrcli = ALiNls()
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elif self.ASRMode == "funasr":
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asrcli = FunASR()
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return asrcli
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def __get_history_average(self, number):
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total = 0
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num = 0
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for i in range(len(self.__history_level) - 1, -1, -1):
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level = self.__history_level[i]
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total += level
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num += 1
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if num >= number:
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break
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return total / num
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def __get_history_percentage(self, number):
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return (self.__get_history_average(number) / self.__MAX_LEVEL) * 1.05 + 0.02
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def __print_level(self, level):
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text = ""
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per = level / self.__MAX_LEVEL
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if per > 1:
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per = 1
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bs = int(per * self.__MAX_BLOCK)
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for i in range(bs):
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text += "#"
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for i in range(self.__MAX_BLOCK - bs):
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text += "-"
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print(text + " [" + str(int(per * 100)) + "%]")
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def reset_wakeup_status(self):
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self.wakeup_matched = False
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def __waitingResult(self, iat: asrclient):
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if self.__fay.playing:
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return
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self.processing = True
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t = time.time()
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tm = time.time()
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# 等待结果返回
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while not iat.done and time.time() - t < 1:
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time.sleep(0.01)
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text = iat.finalResults
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util.log(1, "语音处理完成！ 耗时: {} ms".format(math.floor((time.time() - tm) * 1000)))
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if len(text) > 0:
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if cfg.config['source']['wake_word_enabled']:
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if not self.wakeup_matched:
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#唤醒词判断
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wake_word = cfg.config['source']['wake_word']
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wake_word_list = wake_word.split(',')
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wake_up = False
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for word in wake_word_list:
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if word in text:
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wake_up = True
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if wake_up:
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self.wakeup_matched = True # 唤醒成功
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util.log(1, "唤醒成功！")
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self.on_speaking('唤醒')
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self.processing = False
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self.timer.cancel() # 取消之前的计时器任务
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else:
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util.log(1, "[!] 待唤醒！")
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wsa_server.get_web_instance().add_cmd({"panelMsg": ""})
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else:
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self.on_speaking(text)
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self.processing = False
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self.timer.cancel() # 取消之前的计时器任务
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self.timer = threading.Timer(60, self.reset_wakeup_status) # 重设计时器为60秒
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self.timer.start()
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else:
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self.on_speaking(text)
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self.processing = False
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else:
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if self.wakeup_matched:
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self.wakeup_matched = False
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util.log(1, "[!] 语音未检测到内容！")
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self.processing = False
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self.dynamic_threshold = self.__get_history_percentage(30)
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wsa_server.get_web_instance().add_cmd({"panelMsg": ""})
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if not cfg.config["interact"]["playSound"]: # 非展板播放
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content = {'Topic': 'Unreal', 'Data': {'Key': 'log', 'Value': ""}}
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wsa_server.get_instance().add_cmd(content)
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def __record(self):
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try:
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stream = self.get_stream() #把get stream的方式封装出来方便实现麦克风录制及网络流等不同的流录制子类
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except Exception as e:
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print(e)
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util.log(1, "请检查设备是否有误，再重新启动!")
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return
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isSpeaking = False
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last_mute_time = time.time()
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last_speaking_time = time.time()
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data = None
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while self.__running:
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try:
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data = stream.read(1024, exception_on_overflow=False)
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except Exception as e:
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data = None
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print(e)
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util.log(1, "请检查设备是否有误，再重新启动!")
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return
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if not data:
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continue
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if cfg.config['source']['record']['enabled']:
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if len(cfg.config['source']['record'])<3:
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channels = 1
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else:
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channels = int(cfg.config['source']['record']['channels'])
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#只获取第一声道
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data = np.frombuffer(data, dtype=np.int16)
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data = np.reshape(data, (-1, channels)) # reshaping the array to split the channels
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mono = data[:, 0] # taking the first channel
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data = mono.tobytes()
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level = audioop.rms(data, 2)
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if len(self.__history_data) >= 5:
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self.__history_data.pop(0)
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if len(self.__history_level) >= 500:
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self.__history_level.pop(0)
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self.__history_data.append(data)
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self.__history_level.append(level)
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percentage = level / self.__MAX_LEVEL
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history_percentage = self.__get_history_percentage(30)
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if history_percentage > self.__dynamic_threshold:
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self.__dynamic_threshold += (history_percentage - self.__dynamic_threshold) * 0.0025
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elif history_percentage < self.__dynamic_threshold:
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self.__dynamic_threshold += (history_percentage - self.__dynamic_threshold) * 1
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soon = False
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if percentage > self.__dynamic_threshold and not self.__fay.speaking:
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last_speaking_time = time.time()
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if not self.__processing and not isSpeaking and time.time() - last_mute_time > _ATTACK:
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soon = True #
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isSpeaking = True #用户正在说话
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util.log(3, "聆听中...")
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self.__aLiNls = self.asrclient()
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try:
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self.__aLiNls.start()
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except Exception as e:
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print(e)
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for buf in self.__history_data:
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self.__aLiNls.send(buf)
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else:
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last_mute_time = time.time()
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if isSpeaking:
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if time.time() - last_speaking_time > _RELEASE:
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isSpeaking = False
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self.__aLiNls.end()
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util.log(1, "语音处理中...")
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self.__fay.last_quest_time = time.time()
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self.__waitingResult(self.__aLiNls)
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if not soon and isSpeaking:
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self.__aLiNls.send(data)
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def set_processing(self, processing):
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self.__processing = processing
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def start(self):
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MyThread(target=self.__record).start()
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def stop(self):
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self.__running = False
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self.__aLiNls.end()
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@abstractmethod
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def on_speaking(self, text):
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pass
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#TODO Edit by xszyou on 20230113:把流的获取方式封装出来方便实现麦克风录制及网络流等不同的流录制子类
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@abstractmethod
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def get_stream(self):
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pass
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