• 录音--获取语音流(pyAudio)


    这是学习时的笔记,包含相关资料链接,有的当时没有细看,记录下来在需要的时候回顾。

    有些较混乱的部分,后续会再更新。

    欢迎感兴趣的小伙伴一起讨论,跪求大神指点~


    录音-语音流(pyAudio)

    tags:voice


    进行语音流检测

    pyAudio
    http://old.sebug.net/paper/books/scipydoc/wave_pyaudio.html
    注:这部分是通过录音设备给语音激活检测传输语音流。

    "path/to/vad/audio_stream.py"
    #!usr/bin/env python
    # -*- coding: utf-8 -*-
    
    import numpy as np
    from pyaudio import PyAudio,paInt16
    from datetime import datetime
    import wave
    from Tkinter import *
    import sys
    from ffnn import FFNNVADGeneral
    
    import logging
    # import chardet    # 查看编码
    
    # define of params
    NUM_SAMPLES =160
    FRAMERATE = 16000
    CHANNELS = 1
    SAMPWIDTH = 2
    FORMAT = paInt16
    TIME = 125
    FRAMESHIFT = 160
    
    def save_wave_file(filename,data):
        '''save the date to the wav file'''
        wf = wave.open(filename,'wb')
        wf.setnchannels(CHANNELS)
        wf.setsampwidth(SAMPWIDTH)
        wf.setframerate(FRAMERATE)
        wf.writeframes("".join(data)) # ""中间不能有空格,不然语音录入会有很多中断。
        wf.close()
    
    def my_button(root,label_text,button_text,button_stop,button_func,stop_func):
        '''create label and button'''
        label = Label(root,text=label_text,width=30,height=3).pack()
        button = Button(root,text=button_text,command=button_func,anchor='center',width=30,height=3).pack()
        button = Button(root,text=button_stop,command=stop_func,anchor='center',width=30,height=3).pack()
    
    def record_wave():
        '''open the input of wave'''
        pa = PyAudio()
        # 录音
        stream = pa.open(format=FORMAT,
                         channels=CHANNELS,
                         rate=FRAMERATE,
                         input=True,
                         frames_per_buffer=NUM_SAMPLES)   #一个buffer存NUM_SAMPLES个字节,作为一帧
    
        vad = FFNNVADGeneral('/path/to/VAD/alex-master/alex/tools/vad_train/model_voip/vad_nnt_546_hu32_hl1_hla6_pf10_nf10_acf_1.0_mfr20000_mfl20000_mfps0_ts0_usec00_usedelta0_useacc0_mbo1_bs100.tffnn', 
        			filter_length=2, sample_rate=16000, framesize=512, frameshift=160,
        			usehamming=True, preemcoef=0.97, numchans=26, ceplifter=22, numceps=12, 
        			enormalise=True, zmeansource=True, usepower=True, usec0=False, 
        			usecmn=False, usedelta=False, useacc=False, n_last_frames=10, 
        			n_prev_frames=10, lofreq=125, hifreq=3800, mel_banks_only=True)
        # 语音激活检测神经网络方法的类FFNNVADGeneral.
    
        save_buffer = []
        count = 0
        # logging设置,用于记录日志
        logging.basicConfig(level=logging.INFO,
                            filename='log.txt',
                            filemode ='w',
                            format='%(message)s')
                            
        while count < TIME*4:
            string_audio_data = stream.read(NUM_SAMPLES)	
            result = vad.decide(string_audio_data)
            frame = count*NUM_SAMPLES/float(FRAMESHIFT)
    	time = count*NUM_SAMPLES/float(FRAMERATE)  # time=frame*frameshift/framerate
    	logging.info('frame: '+str(frame)+' time: '+str(time)+' prob: '+str(result))  # logging记录字符串,用‘+’连接
            save_buffer.append(string_audio_data)
            count += 1
            #chardet.detect(string_audio_data)  #查看编码类型
            print "."
       
        filename = datetime.now().strftime("%Y-%m-%d_%H_%M_%S")+".wav"
        save_wave_file(filename,save_buffer)
          
        save_buffer = []
        print "filename,saved."
    
    def record_stop():
        # stop record the wave
        sys.exit(0)
    
    def main():
        root = Tk()
        root.geometry('300x200+200+200')
        root.title('record wave')
        my_button(root,"Record a wave","clik to record","stop recording",record_wave,record_stop)
        root.mainloop()
        
    if __name__ == "__main__":
        main()
    
    
    
    # error
    $ bt_audio_service_open: connect() failed: Connection refused (111)
    # 解决: 貌似有多余蓝牙库却没有蓝牙设备
    $ sudo apt-get purge bluez-alsa
    
    # Warning
    $ ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.rear
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.center_lfe
    ALSA lib pcm.c:2239:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.side
    Cannot connect to server socket err = No such file or directory
    Cannot connect to server request channel
    jack server is not running or cannot be started
    # 是由于usr/share/alsa/alsa.conf默认设置
    
    
    

    录音

    sudo apt-get update   #更新软件源,最好使用国内的软件源,如何配置源参考百度。
    sudo apt-get upgrade  #升级软件包
    sudo apt-get install alsa-utils alsa-tools alsa-tools-gui alsamixergui #安装所需软件包
    
    # 查看音频设备
    $ arecord -l
    > card 0: PCH [HDA Intel PCH], device 0: ALC887-VD Analog [ALC887-VD Analog]
      Subdevices: 1/1
      Subdevice #0: subdevice #0
    card 0: PCH [HDA Intel PCH], device 2: ALC887-VD Alt Analog [ALC887-VD Alt Analog]
      Subdevices: 1/1
      Subdevice #0: subdevice #0
      # 机器有多于一个声卡,可以用下面的命令显示出来
      $ cat /proc/asound/cards
      >  0 [PCH            ]: HDA-Intel - HDA Intel PCH
                          HDA Intel PCH at 0xf7210000 irq 29
     1 [NVidia         ]: HDA-Intel - HDA NVidia
                          HDA NVidia at 0xf7080000 irq 17
     
      # 每一个声卡有一个card number和一个device number,可以用下面命令显示出来
      $ aplay -l
      > card 0: PCH [HDA Intel PCH], device 0: ALC887-VD Analog [ALC887-VD Analog]
      Subdevices: 1/1
      Subdevice #0: subdevice #0
    card 0: PCH [HDA Intel PCH], device 1: ALC887-VD Digital [ALC887-VD Digital]
      Subdevices: 1/1
      Subdevice #0: subdevice #0
    card 1: NVidia [HDA NVidia], device 3: HDMI 0 [HDMI 0]
      Subdevices: 1/1
      Subdevice #0: subdevice #0
    card 1: NVidia [HDA NVidia], device 7: HDMI 1 [HDMI 1]
      Subdevices: 1/1
      Subdevice #0: subdevice #0
    
    # 录音
    $ arecord -D "plughw:0,0" -f S16_LE -r 16000 -d 5 -t wav file.wav
    # -D 选择设备  试过hw:1,0 hw:0,2 只有hw:0,0可以录音
    # -f 录音格式  S16_LE代表有符号16位小端序
    # -r 采样率
    # -t 录音时长
    # file.wav 文件名
    
    # 不添加plug会有警示,因为是外置声卡
    Warning: rate is not accurate (requested = 16000Hz, got = 44100Hz)  please, try the plug plugin 
    
    # 验证录音
    $ aplay file.wav
    
    

    一般alsa设置了一个defaults设备,音频播放软件默认使用defaults设备输出声音。defaults设备定义在alsa.conf中,内容如下:

    [plain]  
    #  
    # defaults  
    #  
      
    # show all name hints also for definitions without hint {} section  
    defaults.namehint.showall off  
    # show just basic name hints  
    defaults.namehint.basic on  
    # show extended name hints  
    defaults.namehint.extended off  
    #  
    defaults.ctl.card 0  
    defaults.pcm.card 0  
    defaults.pcm.device 0  
    defaults.pcm.subdevice -1  
    ……   
    

    defaults会默认匹配card number和device number比较小的声卡。
    如果要修改,则修改/etc/asound.conf或~/.asoundrc。比如我要把defaults匹配到card 1,device 0上,则添加一下几行:

    [plain]  
    $ sudo vim /etc/asound.conf   
    defaults.pcm.card 1  
    defaults.pcm.device 3  
    defaults.ctl.card 1   
    

    c++录音

    https://github.com/aaronaanderson/ofxPortSF


    有些可能记录时忘记记录获取信息的网站地址,有不当之处请指正~~
    (若非特别声明,文章均为Vanessa的个人笔记,转载请注明出处。文章如有侵权内容,请联系我,我会及时删除)

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  • 原文地址:https://www.cnblogs.com/Vanessa-Feng/p/7452108.html
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