WO2016112635A1 - 多声道的数字麦克风 - Google Patents

多声道的数字麦克风 Download PDF

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Publication number
WO2016112635A1
WO2016112635A1 PCT/CN2015/081411 CN2015081411W WO2016112635A1 WO 2016112635 A1 WO2016112635 A1 WO 2016112635A1 CN 2015081411 W CN2015081411 W CN 2015081411W WO 2016112635 A1 WO2016112635 A1 WO 2016112635A1
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array
unit
digital
signal
channel
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PCT/CN2015/081411
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English (en)
French (fr)
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刘鑫
陈裕金
周勇俊
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芋头科技（杭州）有限公司
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Priority to EP15877551.0A priority Critical patent/EP3247130A4/en
Priority to JP2017537989A priority patent/JP2018506228A/ja
Priority to SG11201705717YA priority patent/SG11201705717YA/en
Priority to CA2973246A priority patent/CA2973246C/en
Priority to US14/764,297 priority patent/US10212522B2/en
Priority to NZ734344A priority patent/NZ734344A/en
Priority to KR1020177022421A priority patent/KR20170104566A/ko
Publication of WO2016112635A1 publication Critical patent/WO2016112635A1/zh
Priority to AU2017101065A priority patent/AU2017101065A4/en
Priority to ZA2017/05425A priority patent/ZA201705425B/en

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R19/00—Electrostatic transducers
    • H04R19/04—Microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R29/00—Monitoring arrangements; Testing arrangements
    • H04R29/004—Monitoring arrangements; Testing arrangements for microphones
    • H04R29/005—Microphone arrays
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
    • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
    • H04R2201/401—2D or 3D arrays of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2410/00—Microphones
    • H04R2410/01—Noise reduction using microphones having different directional characteristics
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2420/00—Details of connection covered by H04R, not provided for in its groups
    • H04R2420/09—Applications of special connectors, e.g. USB, XLR, in loudspeakers, microphones or headphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic

Definitions

  • the present invention relates to the field of sound pickup devices, and more particularly to a digital microphone of a multi-channel USB interface.
  • microphones are widely used in various voice communication scenarios, such as stage, large conference rooms, news sites, remote video conferences, voice chat, and voice recognition.
  • the microphone is usually used as an isolated microphone.
  • the sound range of a single microphone is limited, and all sounds (including noise) within the pickup range are received. Therefore, the audio signal received by a single microphone is usually composed of multiple sound sources and the noise superimposed by the surrounding environment. Since the sound source (speaker) may move within a certain range during the output process, and there may be factors such as multipath reflection and reverberation of other sound sources in the sound pickup range, these factors will cause the audio signal received by a single microphone. The signal-to-noise ratio is reduced, resulting in a degradation in audio signal quality.
  • traditional microphones In noise processing, traditional microphones generally use frequency domain for power spectrum cancellation and filtering techniques to suppress noise. However, these received audio signals usually overlap each other in time and frequency spectrum. Therefore, it is quite difficult to separate different audios and effectively suppress noise and interference, and the denoising effect is poor.
  • a multi-channel digital microphone that includes:
  • An array picking unit includes a plurality of sound collecting modules, wherein the plurality of sound collecting modules are arranged in an array, each of the sound collecting modules is configured to pick up an audio signal of a spatial component, and the array picking unit is configured to The audio signals of the spatial components are respectively converted into a plurality of digital signal outputs;
  • a processing unit is coupled to the array pick-up unit for extracting and denoising a plurality of the digital signals.
  • the method further includes:
  • the clock signal including a high level and a low level
  • a channel selection unit is connected to the processing unit for dividing the plurality of digital signals processed by the processing unit into two channels according to the high level and the low level of the clock signal.
  • the method further includes:
  • a transmitting unit is connected to the channel selecting unit for outputting the digital signal.
  • the sound pickup module uses a silicon film to pick up an audio signal of a spatial domain component.
  • the digital signal output by the array pick-up unit is a pulse density modulated signal.
  • the processing unit includes a decimation filter, and the decimation filter is configured to perform low-pass filtering on the digital signal, convert the digital signal into a digital signal of a low-frequency high-order stream, and perform digital filtering.
  • the decimation filter is configured to perform low-pass filtering on the digital signal, convert the digital signal into a digital signal of a low-frequency high-order stream, and perform digital filtering.
  • the sending unit adopts a USB interface module.
  • a sound source localization system comprising the multi-channel digital microphone.
  • the invention adds a spatial domain based on the time domain and the frequency domain to not receive the received space.
  • the signals in the same direction are subjected to pickup and denoising.
  • a plurality of sound pickup modules can be used to form an array in the spatial domain.
  • the array pickup unit can pick up audio signals in multiple directions in the spatial domain; the processing unit can perform denoising on the audio signals to improve the audio quality.
  • FIG. 1 is a block diagram of an embodiment of a multi-channel digital microphone of the present invention.
  • a multi-channel digital microphone includes: an array pickup unit 1 and a processing unit 2; the array pickup unit 1 includes a plurality of sound pickup modules 11, and a plurality of sound pickup modules 11 are arranged in an array, each The sound collecting module 11 is configured to pick up an audio signal of a spatial component, and the array picking unit 1 is configured to convert the audio signals of the plurality of spatial components into a plurality of digital signal outputs respectively; the processing unit 2 is connected to the array picking unit 1 for A plurality of digital signals are extracted and denoised.
  • a plurality of sound collecting modules 11 can be used to form an array in the spatial domain, and the array pickup unit 1 can pick up audio signals in multiple directions in the spatial domain; the array picking unit 1 can adopt multiple numbers.
  • the audio signal of the spatial domain component is converted into a plurality of digital signal outputs, and the air domain The component is determined by the spatial position of the sound pickup module 11 in the array; the processing unit 2 can perform denoising processing on the audio signal to improve the audio quality.
  • the processing unit 2 can adopt a digital signal processing chip (DSP).
  • DSP chip has the advantages of embedability, high-speed performance, good stability, and high precision.
  • the DSP chip can convert the signal of the high-frequency low-order stream into Low frequency high bit stream signal.
  • the method further includes: a clock signal and a channel selection unit 3, the clock signal includes a high level and a low level; the channel selection unit 3 is coupled to the processing unit 2 for high level and low according to the clock signal.
  • the level divides the plurality of digital signals processed by the processing unit 2 into two channels.
  • the received digital signal is divided into a plurality of channels by the channel selecting unit 3 to improve the layering and stereoscopic effects of the audio.
  • it may further include: a transmitting unit 4 connected to the channel selecting unit 3 for outputting the digital signal.
  • the digital signal can be transmitted to the mobile terminal through the transmitting unit 4.
  • the sound pickup module 11 picks up an audio sound pressure signal of a spatial domain component using a silicon film. Silicon film can be used to detect sound pressure.
  • the digital signal output by the array pickup unit 1 is a Pulse Density Modulation (PDM) signal.
  • PDM Pulse Density Modulation
  • the multi-channel digital microphone is equivalent to a microphone array composed of a plurality of pickups, and a spatial domain is added on the basis of the time domain and the frequency domain to process the received signals from different directions in the space.
  • the processing unit 2 may include a Decimator for converting the digital signal into a digital signal of a low frequency high bit stream and performing digital filtering after low pass filtering the digital signal.
  • the decimation filter is downsampled and low-pass filtered, and the high-frequency low-bit stream signal can be converted into a low-frequency high-bit stream PCM signal, and the quantization noise is filtered to realize digital filtering and gain of the digital signal. Adjustment.
  • the transmitting unit 4 can employ a USB interface module.
  • the USB interface module in this embodiment supports hot plugging, and the standard unified can be connected to the mobile terminal, and the digital signal can be transmitted to the mobile terminal at high speed.
  • a sound source localization system comprising a multi-channel digital microphone.
  • the sound source localization system in this embodiment can be applied to a smart home robot.
  • the sound source localization system can receive an audio signal through a multi-channel digital microphone to locate, homing and follow the sound source to move through the multi-channel digital
  • the extraction and separation of the audio signal by the microphone and the denoising can improve the real-time and accuracy of the positioning of the sound source positioning system.
  • the multi-channel digital microphone in this embodiment can also be applied to the self-sound elimination system of the smart home robot.
  • the robot plays music, it will affect the audio recognition function, and the multi-channel digital microphone will have multiple airspaces.
  • the component audio signals are respectively converted into a plurality of digital signals, and the audio signals are sent to an Android device through the USB interface module for self-sound elimination, thereby realizing the robot's own sound source elimination function.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • General Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Obtaining Desirable Characteristics In Audible-Bandwidth Transducers (AREA)

Abstract

本发明公开了多声道的数字麦克风，包括：阵列拾取单元和处理单元，阵列拾取单元包括复数个拾音模块，复数个所述拾音模块呈阵列形式排列，每个所述拾音模块用以拾取一空域分量的音频信号，所述阵列拾取单元用以将复数个空域分量的音频信号分别转换为复数个数字信号输出；处理单元连接所述阵列拾取单元，用以将复数个所述数字信号进行提取和去噪处理。本发明在时域和频域的基础上增加一个空间域来对接收到的来自空间不同方向的信号进行拾音去噪处理。采用复数个拾音模块可构成一在空间域的阵列，通过阵列拾取单元可在空间域内对多个方向的音频信号进行拾取；采用处理单元可对音频信号进行去噪处理提高音频质量。

Description

多声道的数字麦克风 技术领域
本发明涉及拾音装置领域，尤其涉及一种多声道USB接口的数字麦克风。
背景技术
作为语音拾取工具，麦克风被广泛地应用于各种语音通讯场景，如舞台、大型会议室、新闻现场、远程电视会议、语音聊天和语音识别等。
目前麦克风在使用时通常是以孤立的麦克风为主。在实际环境中，单个麦克风的拾音范围有限，在拾音范围内的所有声音(包括噪声)都会被接收。因此，单个麦克风接收的音频信号，通常是由多个声源以及周围环境的噪声叠加组成的。由于声源(说话人)在输出的过程中可能在一定区域内范围移动，同时在拾音范围内可能存在其它音源的多径反射以及混响等因素，这些因素都会使单个麦克风接收的音频信号信噪比降低，从而导致音频信号质量降低。在噪声处理上，传统麦克风一般采用频域进行功率谱抵消和滤波技术来抑制噪声。然而，这些接收到的音频信号通常在时间和频谱上相互重叠，因此，要分离出不同的音频并有效抑制噪声和干扰相当困难，去噪效果差。
发明内容
针对现有的单个麦克风存在的上述问题，现提供一种旨在时域和频域的基础上增加一个空间域来对接收到的来自空间不同方向的信号进行拾音去噪处理的多声道的数字麦克风。
具体技术方案如下：
一种多声道的数字麦克风，包括：
一阵列拾取单元，包括复数个拾音模块，复数个所述拾音模块呈阵列形式排列，每个所述拾音模块用以拾取一空域分量的音频信号，所述阵列拾取单元用以将复数个空域分量的音频信号分别转换为复数个数字信号输出；
一处理单元，连接所述阵列拾取单元，用以将复数个所述数字信号进行提取和去噪处理。
优选的，还包括：
一时钟信号，所述时钟信号包括高电平和低电平；
一声道选择单元，连接所述处理单元，用以根据所述时钟信号的高电平和低电平将经所述处理单元处理后的复数个数字信号分为两个声道。
优选的，还包括：
一发送单元，连接所述声道选择单元，用以将所述数字信号输出。
优选的，所述拾音模块采用硅薄膜拾取一空域分量的音频信号。
优选的，所述阵列拾取单元输出的数字信号为脉冲密度调制信号。
优选的，所述处理单元包括抽取滤波器，所述抽取滤波器用以对所述数字信号进行低通滤波后，将所述数字信号转换为低频高位流的数字信号，并进行数字化滤波。
优选的，所述发送单元采用USB接口模块。
一种声源定位系统，包括所述多声道的数字麦克风。
上述技术方案的有益效果：
本发明在时域和频域的基础上增加一个空间域来对接收到的来自空间不 同方向的信号进行拾音去噪处理。采用复数个拾音模块可构成一在空间域的阵列，通过阵列拾取单元可在空间域内对多个方向的音频信号进行拾取；采用处理单元可对音频信号进行去噪处理提高音频质量。
附图说明
图1为本发明所述多声道的数字麦克风的一种实施例的模块图。
具体实施方式
下面将结合本发明实施例中的附图，对本发明实施例中的技术方案进行清楚、完整地描述，显然，所描述的实施例仅仅是本发明一部分实施例，而不是全部的实施例。基于本发明中的实施例，本领域普通技术人员在没有作出创造性劳动的前提下所获得的所有其他实施例，都属于本发明保护的范围。
需要说明的是，在不冲突的情况下，本发明中的实施例及实施例中的特征可以相互组合。
下面结合附图和具体实施例对本发明作进一步说明，但不作为本发明的限定。
如图1所示，一种多声道的数字麦克风，包括：阵列拾取单元1和处理单元2；阵列拾取单元1包括复数个拾音模块11，复数个拾音模块11呈阵列形式排列，每个拾音模块11用以拾取一空域分量的音频信号，阵列拾取单元1用以将复数个空域分量的音频信号分别转换为复数个数字信号输出；处理单元2连接阵列拾取单元1，用以将复数个数字信号进行提取和去噪处理。
在本实施例中，采用复数个拾音模块11可构成一在空间域的阵列，通过阵列拾取单元1可在空间域内对多个方向的音频信号进行拾取；采用阵列拾取单元1可将复数个空域分量的音频信号转换为复数个数字信号输出，空域 分量是拾音模块11在阵列中的空间位置决定的；采用处理单元2可对音频信号进行去噪处理以提高音频质量。
进一步地，处理单元2可采用数字信号处理芯片(Digital Signal Process，DSP)，DSP芯片具有可嵌入性、高速性能、稳定性好及精度高等优点，DSP芯片可将高频低位流的信号转换为低频高位流的信号。
在优选的实施例中，还可包括：时钟信号和声道选择单元3，时钟信号包括高电平和低电平；声道选择单元3连接处理单元2，用以根据时钟信号的高电平和低电平将经处理单元2处理后的复数个数字信号分为两个声道。
在本实施例中通过声道选择单元3将接收到的数字信号分为多个声道，以提高音频的层次感和立体效果。
在优选的实施例中，还可包括：一发送单元4，发送单元4连接声道选择单元3，用以将数字信号输出。
在本实施例中可通过发送单元4将数字信号发送至移动终端。
在优选的实施例中，拾音模块11采用硅薄膜拾取一空域分量的音频声压信号。硅薄膜可用于检测声压。
在优选的实施例中，阵列拾取单元1输出的数字信号为脉冲密度调制(Pulse Density Modulation，PDM)信号，PDM信号的幅度变化越剧烈，脉冲密度越密。
在本实施例中多声道的数字麦克风相当于是由多个拾音器构成的麦克风阵列，在时域和频域的基础上增加一个空间域来对接收到的来自空间不同方向的信号进行处理。通过将拾取的音频信号转换为PDM信号，并输出给DSP芯片进行处理，采用数字化技术可有效地解决声音分离及噪声抑制的问题，从而提高拾取音频信号的能力。
在优选的实施例中，处理单元2可包括抽取滤波器(Decimator)，抽取滤波器用以对数字信号进行低通滤波后，将数字信号转换为低频高位流的数字信号，并进行数字化滤波。
在本实施例中，在抽取滤波器下采样并进行低通滤波，可将高频低位流的信号转换成低频高位流的PCM信号，同时滤除量化噪声，实现对数字信号的数字化滤波和增益调节。
在优选的实施例中，发送单元4可采用USB接口模块。
在本实施例中的USB接口模块支持热插拔、标准统一可以与移动终端连接，可将数字信号高速传输至移动终端。
一种声源定位系统，包括多声道的数字麦克风。
本实施例中的声源定位系统可以应用于智能家居机器人上，声源定位系统可通过多声道的数字麦克风接收音频信号对音源进行定位、寻向并跟随音源移动，通过多声道的数字麦克风对音频信号的提取分离以及去噪可以提高声源定位系统定位的实时性与准确性。
在本实施例中的多声道的数字麦克风还可应用在智能家居机器人的自身音源消除系统上，当机器人播放音乐时会对音频识别功能造成影响，通过多声道的数字麦克风将复数个空域分量的音频信号分别转换为复数个数字信号，并通过USB接口模块将音频信号发送至安卓(Android)设备进行自身音源消除，以实现机器人的自身音源消除功能。
以上所述仅为本发明较佳的实施例，并非因此限制本发明的实施方式及保护范围，对于本领域技术人员而言，应当能够意识到凡运用本发明说明书及图示内容所作出的等同替换和显而易见的变化所得到的方案，均应当包含在本发明的保护范围内。

Claims (8)

  1. 一种多声道的数字麦克风，其特征在于，包括：
    一阵列拾取单元，包括复数个拾音模块，复数个所述拾音模块呈阵列形式排列，每个所述拾音模块用以拾取一空域分量的音频信号，所述阵列拾取单元用以将复数个空域分量的音频信号分别转换为复数个数字信号输出；
    一处理单元，连接所述阵列拾取单元，用以将复数个所述数字信号进行提取和去噪处理。
  2. 如权利要求1所述多声道的数字麦克风，其特征在于，还包括：
    一时钟信号，所述时钟信号包括高电平和低电平；
    一声道选择单元，连接所述处理单元，用以根据所述时钟信号的高电平和低电平将经所述处理单元处理后的复数个数字信号分为两个声道。
  3. 如权利要求2所述多声道的数字麦克风，其特征在于，还包括：
    一发送单元，连接所述声道选择单元，用以将所述数字信号输出。
  4. 如权利要求1所述多声道的数字麦克风，其特征在于，所述拾音模块采用硅薄膜拾取一空域分量的音频信号。
  5. 如权利要求1所述多声道的数字麦克风，其特征在于，所述阵列拾取单元输出的数字信号为脉冲密度调制信号。
  6. 如权利要求1所述多声道的数字麦克风，其特征在于，所述处理单元包括抽取滤波器，所述抽取滤波器用以对所述数字信号进行低通滤波后，将所述数字信号转换为低频高位流的数字信号，并进行数字化滤波。
  7. 如权利要求3所述多声道的数字麦克风，其特征在于，所述发送单元采用USB接口模块。
  8. 一种声源定位系统，其特征在于，包括如权利要求1至7所述多声道的数字麦克风。
PCT/CN2015/081411 2015-01-12 2015-06-12 多声道的数字麦克风 WO2016112635A1 (zh)

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