WO2012069020A1 - 语音增强方法、装置及头戴式降噪通信耳机 - Google Patents

语音增强方法、装置及头戴式降噪通信耳机 Download PDF

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Publication number
WO2012069020A1
WO2012069020A1 PCT/CN2011/082993 CN2011082993W WO2012069020A1 WO 2012069020 A1 WO2012069020 A1 WO 2012069020A1 CN 2011082993 W CN2011082993 W CN 2011082993W WO 2012069020 A1 WO2012069020 A1 WO 2012069020A1
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Prior art keywords
noise
signal
speech
microphone
sound signal
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PCT/CN2011/082993
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English (en)
French (fr)
Inventor
赵剑
刘崧
李波
华洋
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歌尔声学股份有限公司
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Application filed by 歌尔声学股份有限公司 filed Critical 歌尔声学股份有限公司
Priority to JP2013506486A priority Critical patent/JP5635182B2/ja
Priority to KR1020127028284A priority patent/KR101500823B1/ko
Priority to DK11843100.6T priority patent/DK2555189T3/en
Priority to EP11843100.6A priority patent/EP2555189B1/en
Priority to US13/637,715 priority patent/US9240195B2/en
Publication of WO2012069020A1 publication Critical patent/WO2012069020A1/zh

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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
    • G10L2021/02165—Two microphones, one receiving mainly the noise signal and the other one mainly the speech signal
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/08—Mouthpieces; Microphones; Attachments therefor
    • H04R1/083—Special constructions of mouthpieces
    • 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
    • H04R2201/10—Details of earpieces, attachments therefor, earphones or monophonic headphones covered by H04R1/10 but not provided for in any of its subgroups
    • H04R2201/107—Monophonic and stereophonic headphones with microphone for two-way hands free communication
    • 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
    • H04R2410/05—Noise reduction with a separate noise microphone

Definitions

  • the present invention relates to the field of voice signal processing technologies, and in particular, to a voice enhancement method and apparatus for a voice transmitting terminal and a headphone noise reduction communication earphone.
  • the communication and communication between people is becoming faster and more convenient, and the wide application of various communication devices and technologies greatly facilitates people's lives and Improve work efficiency.
  • the noise problem that comes with the development of society also seriously affects the clarity and intelligibility of communication voice.
  • the noise is high to a certain extent, communication can not be carried out at all, and it will hurt people's hearing and Physical and mental health.
  • the real-time communication and the clarity and intelligibility of communication voice are very high.
  • the intensity of external noise is often It will reach more than 100 decibels.
  • the voice enhancement methods of the communication terminals of the commonly used communication devices include two categories, one is to pick up signals by using one or more ordinary microphones, and then the acoustic signal processing method is used to achieve the purpose of speech enhancement; the other is to use special acoustic microphones.
  • the microphone and the vibration microphone are used to achieve the purpose of effectively picking up the voice signal and suppressing the noise.
  • Single microphone speech enhancement is generally referred to as single-channel spectral subtraction speech enhancement technology (see Chinese Invention Patent Application Publication No. CN1684143A, CN101477800A).
  • This technique generally estimates the energy of noise in the current speech by analyzing historical data, and then passes the spectrum.
  • the subtraction method eliminates the noise in the speech to achieve the purpose of speech enhancement.
  • a microphone array speech enhancement technique using two or more microphones is generally a signal received by a microphone as a reference signal, and is estimated in real time by adaptive filtering. The noise component in the other microphone pickup signal is cancelled, and the speech component is preserved, thereby achieving the purpose of speech enhancement.
  • the speech enhancement method using single or multiple ordinary microphones relies heavily on the detection and judgment of the speech state. Otherwise, not only can the noise not be well eliminated, but also the speech signal is greatly damaged. In a low-noise environment, the detection and judgment of the speech state is feasible and accurate, but in a noisy environment, the speech signal will be completely submerged by noise. In this extremely low signal-to-noise ratio, a normal microphone is used. The speech enhancement technology will not get good results or it will not be applicable at all.
  • the other is to use some special acoustic microphones, such as microphones, vibrating microphones, etc., to improve the pickup signal-to-noise ratio in noisy environments, thus achieving the purpose of speech enhancement.
  • the microphone is also called the noise reduction microphone. It is a microphone designed with the principle of differential pressure. It has directivity and “near-talk effect”. It has about 15dB noise reduction effect on noise, especially far-field low-frequency noise.
  • the traffic headset and some professional communication fields use more microphones.
  • the vibrating microphone needs to be well coupled with the vibrating surface to pick up the useful signal, and the noise signal transmitted by the air has a noise reduction effect of 20 to 30 dB.
  • the noise reduction of the microphone is limited and can not effectively suppress the wind noise;
  • the vibration microphone (see Chinese utility model patent specification CN2810077Y) has a noise reduction of 20 to 30 dB in the whole frequency band for noise (including wind noise), but its frequency Poor characteristics, can not effectively pick up the high-frequency information of the voice, the naturalness and intelligibility of the call voice can not be guaranteed, so these two special types of acoustic microphones can not be better used in communication headphones under high-intensity noise environment.
  • an object of the present invention is to provide a speech enhancement scheme capable of effectively combining a vibration microphone and an acoustic signal processing technique for improving a speech signal to noise ratio and a speech quality of a communication transmitter in a high-intensity noise environment.
  • the invention discloses a speech enhancement device, which comprises: an acoustic speech enhancement unit and an electronic speech enhancement unit; wherein
  • the acoustic speech enhancement unit includes: a main vibration microphone and a secondary vibration microphone having a specific relative positional relationship; the specific relative positional relationship causes the main vibration microphone to pick up a voice signal of the user transmitted through the coupled vibration mode and propagated from the air.
  • the external environmental noise signal, the auxiliary vibration microphone mainly picks up the external environmental noise signal propagating from the air, and the external environmental noise signal picked up by the main vibration microphone and the auxiliary vibration microphone from the air has correlation;
  • the electronic speech enhancement unit includes: a voice detection module, an adaptive filtering module, and a post-processing module; wherein
  • a voice detection module configured to determine an update speed of the adaptive filter module according to a sound signal output by the main vibration microphone and the auxiliary vibration microphone, and output a control parameter
  • An adaptive filtering module configured to perform noise reduction filtering on the sound signal output by the main vibration microphone according to the sound signal output by the auxiliary vibration microphone and the control parameter output by the voice detection module, and output the noise-reduced filtered voice signal;
  • the post-processing module is configured to perform further noise reduction and speech high-frequency enhancement processing on the noise-reduced filtered voice signal output by the adaptive filtering module.
  • the invention also discloses a head-mounted noise reduction communication earphone, which comprises a voice signal transmission port and a voice enhancement device as described above;
  • the voice signal transmission port is configured to receive the voice signal after the noise reduction of the voice enhancement device, and transmit the voice signal to the remote user.
  • the invention also discloses a speech enhancement method, the method comprising:
  • the first sound signal includes a voice signal of the user transmitted from the coupled vibration mode and from the air Propagating a recent external environmental noise signal
  • the second sound signal is mainly an external environmental noise signal propagating from the air, and the external environmental noise signal in the first sound signal and the second sound signal have a correlation
  • the speech of the transmitting end is separately enhanced in speech at the acoustic level and the electronic level.
  • using a main vibration microphone and a secondary vibration microphone having a specific relative positional relationship respectively picking up a first voice signal including a user's voice signal and an external environmental noise signal, and mainly using external environmental noise signals
  • the second sound signal because of the vibration microphone structure, can attenuate the external noise by 20 to 30 dB at the time of picking, and the external sound of the first sound signal and the second sound signal have a high correlation, which is an electron.
  • the speech enhancement algorithm on the layer provides a better noise reference signal; firstly, based on the first sound signal and the second sound signal, a control parameter for controlling the update speed of the adaptive filter is determined on the electronic level, and then according to the second sound signal and The control parameter performs noise reduction filtering on the first sound signal to obtain a voice signal with high signal-to-noise ratio, and finally performs further noise reduction and voice high-frequency enhancement processing on the noise-reduced filtered voice signal, thereby greatly improving the The intelligibility and clarity of the voice of the voice. It can be seen that through the above-mentioned acoustic level and electronic level speech enhancement processing, it is finally possible to provide up to 40 to 50 dB of noise reduction at the transmitting end of the communication.
  • the voice signal-to-noise ratio of the communication terminal is greatly improved, and the naturalness and intelligibility of the voice of the voice-transmitting terminal are improved, and the voice signal-to-noise ratio and voice quality in the high-intensity noise environment are greatly improved.
  • FIG. 1 is a schematic structural view of a vibration microphone composed of a microphone with a rubber sleeve
  • FIG. 2 is a schematic structural view of a main and auxiliary vibration microphone mounted on a strut in a speech enhancement device according to the present invention
  • 3A is a schematic view showing a coupling position of a main vibration microphone and a head wearer head
  • 3B is a schematic view showing the effect of coupling the earphone with the microphone struts and the wearer's ankle portion to which the present invention is applied;
  • FIG. 4 is a system block diagram of an electronic layer speech enhancement in the present invention.
  • FIG. 5 is a schematic flowchart diagram of a voice enhancement method according to the present invention.
  • FIG. 6 is a block diagram of a speech enhancement apparatus of the present invention.
  • Figure 7 is a block diagram of a head-mounted noise reduction communication earphone of the present invention.
  • the speech enhancement method of the present invention comprises two major parts.
  • the first part is a speech signal enhancement on the acoustic level, and a main signal with a good signal to noise ratio for the speech enhancement algorithm on the electronic level and a noise reference highly correlated with the main signal.
  • the second part is to use the acoustic signal processing method to further perform speech enhancement processing on the signal, improve the signal-to-noise ratio of the speech, and improve the intelligibility and comfort of the speech at the sending end.
  • the speech enhancement techniques at the acoustic and electronic levels are described separately below.
  • the present invention employs a double vibrating microphone structure.
  • the main vibrating microphone and the auxiliary vibrating microphone have similar structures and are close to each other in spatial position, that is, the main vibrating microphone and the auxiliary vibrating microphone have a specific relative positional relationship.
  • the specific relative positional relationship causes the main vibration microphone to pick up the voice signal of the user transmitted through the coupled vibration mode and the ambient environmental noise signal propagated from the air, and the auxiliary vibration microphone mainly picks up the external environmental noise signal propagating from the air.
  • the ambient noise signals propagating from the air into the main vibration microphone and the auxiliary vibration microphone, respectively, are correlated.
  • the main vibration microphone is in direct contact with the earphone wearer, and the voice signal of the earphone wearer is effectively picked up by coupling vibration, and the auxiliary vibration microphone does not directly contact the earphone wearer, and does not couple the voice signal transmitted through the vibration.
  • the primary and secondary vibrating microphones will have an attenuation of about 20 to 30 dB, and by adjusting the positions of the main and auxiliary microphones, the noise signals picked up by the two vibrating microphones have a good correlation.
  • a microphone having a hermetic rubber sleeve structure is employed as the vibration microphone.
  • 1 is a schematic view showing a structure in which a microphone is placed in a closed rubber sleeve to form a vibration microphone.
  • a microphone (MIC) 10 is placed in the sealed rubber sleeve 20, and is retained between the diaphragm of the microphone 10 and the rubber sleeve 20.
  • a certain air cavity 30 is sealed for sound signals to pass.
  • the external environmental noise propagating from the air can be picked up by the diaphragm of the microphone 10 due to the attenuation of the rubber sleeve 20, so the noise is greatly reduced; and for the vibration signal coupled to the upper surface of the rubber sleeve 20, the rubber sleeve 20
  • the vibration of the surface directly causes a change in the volume of the closed air chamber 30, thereby causing vibration of the diaphragm of the microphone 10, so that the vibration signal of the upper surface of the rubber sleeve 20 is effectively picked up by the microphone 10.
  • the microphone 10 with the rubber sleeve 20 must effectively couple the voice signal of the earphone wearer while isolating the external noise.
  • many parts of the human head will contain certain voice vibration signals (especially low frequency information). Among them, the voice spectrum information contained in the throat and ankle vibration is richer. Therefore, in consideration of the convenience and aesthetics of the wearing of the earphone, in a preferred embodiment of the present invention, the microphone struts shown in FIG. 2 are designed, and a microphone with a rubber sleeve is placed on each of the front and back sides of the struts head.
  • FIG. 3A shows a schematic positional view of the coupling of the main vibration microphone and the head, including the crown 301, the forehead 302, the crotch portion 303, and the ankle portion.
  • Fig. 3B The front side of the rubber sleeve of the main vibration microphone 112 is well coupled with the crotch portion of the earphone wearer, so that the voice information of the earphone wearer can be better picked up.
  • the auxiliary vibration microphone 114 is not directly coupled to the human face, so it is not sensitive to the earphone wearer's voice signal.
  • the main vibration microphone 112 picks up a good speech signal and is attenuated by about 20 ⁇ .
  • the external noise signal of 30dB, the auxiliary vibration microphone 114 picks up the external noise signal which is attenuated by about 20 ⁇ 30dB, and the pure external noise signal picked up by the auxiliary vibration microphone 114 can provide the noise reduction for the next electronic level.
  • Good external noise reference signal is provided.
  • the main vibration microphone 112 and the auxiliary vibration microphone 114 are relatively close, and have a similar rubber sleeve structure, so as to ensure that the external noise signals leaking into the two rubber sleeves have a good correlation to ensure that the electronic layer faces the noise signal. Can be further reduced.
  • auxiliary vibrating microphone 114 in order to prevent the auxiliary vibrating microphone 114 from picking up more vibrating speech signals, thereby causing the speech signal in the main vibrating microphone 112 to be damaged at the electronic level, it is preferable to take a better relationship between the main vibrating microphone 112 and the auxiliary vibrating microphone 114. Vibration isolation treatment measures. In a preferred embodiment of the present invention, some spacers are added between the main and auxiliary microphone sleeves to achieve vibration isolation.
  • the technique of acoustic signal processing is used to further improve the signal-to-noise ratio of the speech signal and improve the naturalness and sharpness of the speech signal picked up by the vibration.
  • the vibration microphone in the present invention is not limited to the above-mentioned microphone with a closed rubber sleeve structure, and the existing bone conduction microphone can be used, or a common electret (ECM) microphone can be used to add a special acoustic structure design. Achieve the effect of a vibrating microphone.
  • ECM electret
  • FIG. 4 is a block diagram of a system for electronic level speech enhancement of signals enhanced by acoustic level speech.
  • the voice enhancement at the electronic level mainly includes a voice detection module 210, an adaptive filtering module 220, and a post-processing module 230, wherein the voice detection module 210 is configured to output sounds according to the primary vibration microphone 112 and the secondary vibration microphone 114.
  • the signal determines the update speed of the adaptive filter module 220 and outputs the control parameter ⁇ ; the adaptive filter module 220 outputs the sound signal to the main vibration microphone 112 according to the sound signal output by the auxiliary vibration microphone 114 and the control parameter ⁇ output by the voice detection module 210.
  • the noise reduction filtering is performed, and the noise-reduced speech signal is output; the post-processing module 230 is configured to perform further noise reduction and speech high-frequency enhancement processing on the noise-reduced filtered speech signal output by the adaptive filtering module 220.
  • the main vibration microphone 112 directly couples the vibration of the wearer's ankle to pick up a larger voice signal; although the secondary vibration microphone 114 is not directly coupled to the ankle, because it is closer to the mouth of the wearer, When the wearer speaks loudly, the voice signal picked up by the auxiliary vibration microphone 114 through the air leak cannot be ignored.
  • the signal of the auxiliary vibration microphone 114 is directly used as the filter reference signal to update the adaptive filter and filter, it is possible to cause damage to the speech, so it must first be used by the speech detection module 210 according to the main vibration microphone 112 and the auxiliary vibration microphone.
  • the sound signal outputted by 114 determines the update speed of the adaptive filter in the adaptive filtering module 220, and outputs a control parameter ⁇ indicating that the update speed of the adaptive filter 221 is controlled.
  • the value of the control parameter ⁇ is determined by calculating the statistical energy ratio P_ratio of the primary vibration microphone 112 and the secondary vibration microphone 114 in the low frequency range, and the larger the energy ratio P_ratio is the main vibration microphone 112.
  • the smaller the energy ratio P_ratio is, the sound that the main vibration microphone 112 picks up.
  • the smaller the proportion of the target speech in the signal the larger the proportion of the ambient noise, the larger the value of ⁇ , and the faster the update speed of the adaptive filter 221 is.
  • the low frequency range refers to a frequency range below 500 Hz.
  • the value range of ⁇ is 0 ⁇ ⁇ ⁇ 1.
  • the adaptive filtering module 220 includes an adaptive filter 221 and a subtractor 222.
  • an FIR filter having a step length P (P ⁇ 1) is used as an adaptive for the noise reduction filter.
  • the sound signals picked up and output by the main vibration microphone 112 and the auxiliary vibration microphone 114 are the first sound signal s1(n) and the second sound signal s2(n), respectively, and the input signal of the adaptive filter 221 is the auxiliary vibration microphone 114.
  • the picked up sound signal s2(n) under the control of the update speed of the control parameter ⁇ , the adaptive filter 221 filters the output signal s3(n), and the subtractor 222 picks up the sound picked up by s3(n) and the main vibration microphone 112.
  • the signal s1(n) is subtracted to obtain a noise canceled signal y(n), and y(n) is fed back to the adaptive filter 221 for re-updating the filter weight.
  • the update speed of the adaptive filter 221 is controlled by the control parameter ⁇ .
  • the adaptive filter 221 quickly converges to the noise from the auxiliary vibration microphone 114.
  • the transfer function H_noise to the main vibration microphone 112 is such that s3(n) is the same as s1(n), and the offset y(n) is small, thereby eliminating noise.
  • the adaptive filter stops updating, so that the adaptive filter does not converge to the speech from the auxiliary vibration microphone 114 to the main vibration microphone 112.
  • the transfer function H_speech, s3(n) is different from s1(n), so that the subtracted speech components are not cancelled, and the output y(n) retains the speech components.
  • 0 ⁇ ⁇ ⁇ 1 that is, the sound signal picked up by the main vibration microphone 112 has both a speech component and an environmental noise component, and the update speed of the adaptive filter 221 is controlled by the number of speech components and environmental noise components. To ensure that noise is removed while retaining speech components.
  • the transfer function H_noise of the noise from the auxiliary vibration microphone 114 to the main vibration microphone 112 is similar to the transfer function H_speech of the speech from the sub-microphone 114 to the main vibration microphone 112, even if the adaptive filter 221 converges to H_noise, The speech causes a certain degree of damage, so it is necessary to use ⁇ to constrain the weight of the adaptive filter 221.
  • time domain adaptive filter is used for noise reduction in the above specific embodiments, those skilled in the art should understand that the filter used in filtering is not limited to the time domain adaptive filter.
  • the frequency domain (subband) adaptive filter can also be used for noise reduction, and the control parameter ⁇ i of each frequency subband can be further obtained by the statistical energy ratio P_ratioi of each frequency subband of the primary vibration microphone 112 and the secondary vibration microphone 114. And independently control the update of each frequency subband of the frequency adaptive filter.
  • i is the identifier of the frequency sub-band, wherein the larger the statistical energy ratio of each frequency sub-band, the smaller the value of ⁇ i corresponding to the frequency sub-band, the value range of ⁇ i is 0 ⁇ ⁇ i ⁇ 1, that is, the index of ⁇ i The range is 0 to 1.
  • the post-processing module 230 includes a single-channel noise reduction sub-module 231 and a speech high-frequency enhancement sub-module 232.
  • the single-channel noise reduction sub-module 231 first calculates the energy of the stationary noise remaining in the output signal y(n) of the adaptive filtering module 220 according to the characteristics of the noise smoothness; in addition, since the high-frequency energy of the voice signal picked up by the vibration mode is small, The clarity and intelligibility of the processed speech are not high. Therefore, the speech high frequency enhancement sub-module 232 is used to enhance the high-frequency component of the speech signal after single-channel noise reduction processing by the single-channel noise reduction sub-module 231. , thereby greatly improving the clarity and intelligibility of the output speech signal, so that the user obtains a sufficiently clear speech signal.
  • the single-channel noise reduction sub-module 231 calculates the noise energy by using a smooth averaging method, and subtracts the noise energy from the signal y(n), thereby further reducing the adaptive filtering module 220.
  • the noise component in the output y(n) retains the speech component therein to achieve the effect of improving the signal-to-noise ratio of the speech signal.
  • FIG. 5 is a schematic flowchart of a specific method for voice enhancement provided by the present invention.
  • the speech enhancement method of the present invention includes the following steps:
  • step S510 the first sound signal s1(n) and the second sound signal s2(n) are respectively picked up by the main vibration microphone 112 and the auxiliary vibration microphone 114, wherein the first sound signal s1(n) includes vibration through coupling
  • the second sound signal s2(n) is mainly an external environmental noise signal leaking from the rubber sleeve into the microphone, and due to the position of the vibration microphone Providing correlation between external environmental noise signals in the first sound signal s1(n) and the second sound signal s2(n);
  • step S520 the update speed of the adaptive filter is determined according to the first sound signal s1(n) and the second sound signal s2(n) and the control parameter ⁇ is output, 0 ⁇ 1;
  • step S530 the first sound signal s1(n) is subjected to noise reduction processing according to the first sound signal s1(n), the second sound signal s2(n), and the control parameter ⁇ by using an adaptive filter;
  • step S550 the high-frequency component is enhanced by the above-described sound signal after eliminating the energy of the residual stationary noise.
  • the above voice enhancement method of the present invention is implemented by a combination of software and hardware.
  • FIG. 6 is a diagram showing the logical structure of a speech enhancement apparatus corresponding to the above-described speech enhancement method of the present invention.
  • the speech enhancement apparatus 600 provided by the present invention includes an acoustic speech enhancement unit 610 and an electronic speech enhancement unit 620.
  • the acoustic speech enhancement unit 610 includes a primary vibration microphone 112 and a secondary vibration microphone 114.
  • the main vibration microphone 112 is configured to pick up a voice signal of a user transmitted through the coupled vibration mode and an ambient noise signal propagated from the air;
  • the auxiliary vibration microphone 114 is configured to pick up an ambient noise signal propagating from the air; and
  • the ambient environmental noise signals propagating from the air into the primary vibration microphone 112 and the secondary vibration microphone 114, respectively, are correlated.
  • the electronic voice enhancement unit 620 includes a voice detection module 210, an adaptive filtering module 220, and a post-processing module 230, wherein the voice detection module 210 is configured to determine the self according to the sound signals output by the primary vibration microphone 112 and the secondary vibration microphone 114. Adapting the update speed of the filtering module 220 and outputting the control parameter ⁇ ; the adaptive filtering module 220 outputs the sound signal output by the auxiliary vibration microphone 114 and the control parameter ⁇ output by the voice detecting module 210 to the main vibration microphone 112.
  • the sound signal is subjected to noise reduction filtering, and the noise-reduced filtered voice signal is output; the post-processing module 230 is configured to further reduce noise and voice high on the noise-reduced filtered voice signal output by the adaptive filtering module 220. Frequency enhancement processing.
  • the speech detecting module 210 is configured to calculate a statistical energy ratio of the sound signal output by the main vibration microphone 112 and the sound signal output by the auxiliary vibration microphone 114 in the low frequency range. Determining the control parameter of the adaptive filter 221; wherein the larger the statistical energy ratio, the smaller the value of the control parameter, and the value range of the control parameter is 0 to 1;
  • the speech detecting module 210 is configured to calculate the statistical energy in each frequency subband by calculating the sound signal output by the main vibrating microphone 112 and the sound signal output by the auxiliary vibrating microphone 114.
  • the ratio determines the control parameter ⁇ i of each frequency subband; wherein the larger the statistical energy ratio of the frequency subband, the smaller the value of the control parameter ⁇ i corresponding to the frequency subband, and the control parameter ⁇ i corresponding to each frequency subband The value ranges from 0 to 1.
  • Figure 7 shows a block diagram of a head mounted noise reduction communication headset 700 having a speech enhancement device in accordance with the present invention.
  • the head-mounted noise reduction communication headset 700 includes a voice signal transmission port 701 and the voice enhancement device 600 as shown in FIG. 6, wherein the voice signal transmission port 701 is configured to transmit a near-end voice signal.
  • the voice signal after the noise reduction by the voice enhancement device 600 is received, and then sent to the remote user by wire or wireless.
  • the functions of the various components of the speech enhancement device 600 and their descriptions are identical to those described above with respect to Figures 4 and 6, and will not be described again.
  • the present invention can eliminate environmental noise from the acoustic level and the electronic level, and greatly improve the speech signal to noise ratio and voice quality under high-intensity noise environment as follows:
  • the double vibration microphone can effectively isolate the noise from the outside air; and for the noise coming in, because the main and auxiliary vibration microphones have similar structure and close spatial position, they leak into the main and auxiliary vibration microphones.
  • the external noise signal has a good correlation.
  • the main vibration microphone can better pick up the earphones.
  • the vibrating voice signal, while the auxiliary vibrating microphone can only pick up the incoming speech signal.
  • the invention is insensitive to the directionality and position of the noise, and has stable noise reduction for the noise in all directions of the near and far fields, and It also has a good noise reduction effect on wind noise.

Abstract

本发明公开了一种语音增强方法、装置及头戴式降噪通信耳机。在本发明的方案中，利用具有特定相对位置关系的主振动传声器和辅振动传声器，分别拾取通过耦合振动方式传过来的使用者的语音信号和从空气中传播进来的外界环境噪声信号的第一声音信号，和拾取主要从空气中传播进来的外界环境噪声信号的第二声音信号，且两振动传声器拾取的外界环境噪声信号具有相关性；根据第一声音信号和第二声音信号确定控制自适应滤波器更新速度的控制参数；根据第二声音信号和控制参数对第一声音信号进行降噪滤波；对降噪滤波后的语音信号做进一步的降噪和语音高频增强处理。本发明的技术方案，能够有效提升高强度噪声环境下的语音信噪比和语音质量。

Description

语音增强方法、装置及头戴式降噪通信耳机
技术领域
本发明涉及语音信号处理技术领域，更为具体地，涉及一种送话端的语音增强方法、装置和头戴式降噪通信耳机。
发明背景
随着技术的进步和社会信息化程度的提高，人与人之间的通信交流方式也越来越快捷和方便，各种各样的通信设备和技术的广泛应用极大地方便了人们的生活和提高了工作效率。但是，伴随社会的发展而随之产生的噪声问题也严重影响到通信语音的清晰度和可懂度，当噪声高到一定程度时，不但通信根本就无法进行，而且会伤害到人的听力和身心健康。尤其是在一些特殊的地方，如机场、车站、大型工业工厂车间等场合，对通信的实时性和通信语音的清晰度和可懂度要求非常高，然而在这些特殊场合，外界噪声的强度往往都会达到100分贝以上，在这种极限噪声情况下进行送话，远端用户接受到的语音信号会完全被环境噪声淹没，根本得不到任何有用的信息。因此有必要在通信设备的送话端采取有效的语音增强方法来提高送话端语音的信噪比。
目前常用的通信设备送话端的语音增强方法包括两大类，一类是采用单个或多个普通麦克风拾取信号，然后采用声学信号处理方法来达到语音增强的目的；另一类是采用特殊声学传声器，如近讲麦克风和振动传声器来达到有效拾取语音信号和抑制噪声的目的。
单个麦克风语音增强一般称之为单通道谱减语音增强技术（参见中国发明专利申请公开说明书CN1684143A，CN101477800A），这种技术一般通过对历史数据的分析来估计当前语音中噪声的能量，然后通过频谱相减的方法消除语音中的噪声来达到语音增强的目的。采用两个或多个麦克风组成的麦克风阵列语音增强技术（参见中国发明专利申请公开说明书CN101466055A，CN1967158A）则通常是用一个麦克风接收到的信号作为参考信号，通过自适应滤波的方法实时的估计并抵消另外一个麦克风拾取信号中的噪声成分，保留语音成分，从而达到语音增强的目的。采用单个或多个普通麦克风的语音增强方法，其性能很大程度上依赖于对语音状态的检测和判断，否则不但不能很好的消除噪声，而且还会给语音信号带来较大的损伤。在低噪声环境中，对语音状态的检测和判断是可行和准确的，但在强噪声环境中，语音信号将会完全被噪声所淹没，在这种极低信噪比情况下，采用普通麦克风的语音增强技术将得不到较好效果或者根本无法适用。
另一类是采用一些特殊的声学传声器，如近讲麦克风、振动传声器等，以在噪声环境下提高拾取语音信噪比，从而达到语音增强的目的。近讲麦克风又称之为降噪麦克风，是采用压差原理进行设计的麦克风，具有指向性和“近讲效应”，对噪声尤其是远场低频噪声有大约15dB左右的降噪效果，现在一般的话务耳机和一些专业通信领域的耳机较多采用近讲麦克风。振动传声器需要与振动面有较好耦合来拾取有用信号，对空气传导过来的噪声信号则有20～30dB的降噪效果。但近讲麦克风的降噪量有限并且不能有效抑制风噪声；振动传声器（参见中国实用新型专利说明书CN2810077Y）虽然对噪声（包括风噪声）在全频带有20～30dB的降噪量，但其频响特性差，不能有效拾取语音的高频信息，通话语音的自然度和可懂度不能保证，因此这两类特殊的声学传声器都不能较好应用于高强度噪声环境下的通信耳机。
发明内容
鉴于上述问题，本发明的目的是提供一种能够有效结合振动传声器和声学信号处理技术的语音增强方案，用来提升高强度噪声环境下通信送话端的语音信噪比和语音质量。
本发明公开了一种语音增强装置，该装置包括：声学语音增强单元和电子语音增强单元；其中，
声学语音增强单元包括：具有特定相对位置关系的主振动传声器和辅振动传声器；所述特定相对位置关系使得主振动传声器拾取通过耦合振动方式传过来的使用者的语音信号和从空气中传播进来的外界环境噪声信号，辅振动传声器主要拾取从空气中传播进来的外界环境噪声信号，并且主振动传声器和辅振动传声器所拾取的从空气中传播进来的外界环境噪声信号具有相关性；
电子语音增强单元包括：语音检测模块、自适应滤波模块和后处理模块；其中，
语音检测模块，用于根据所述主振动传声器和辅振动传声器输出的声音信号确定所述自适应滤波模块的更新速度并输出控制参数；
自适应滤波模块，用于根据所述辅振动传声器输出的声音信号和所述语音检测模块输出的控制参数对所述主振动传声器输出的声音信号进行降噪滤波，并输出降噪滤波后的语音信号；
后处理模块，用于对所述自适应滤波模块输出的降噪滤波后的语音信号做进一步的降噪和语音高频增强处理。
本发明还公开了一种头戴式降噪通信耳机，该通信耳机包括语音信号传送端口和如上所述的语音增强装置；
所述语音信号传送端口，用于接收所述语音增强装置降噪后的语音信号，并传送给远端用户。
本发明还公开了一种语音增强方法，该方法包括：
利用具有特定相对位置关系的主振动传声器和辅振动传声器，分别拾取第一声音信号和第二声音信号；所述第一声音信号包括通过耦合振动方式传过来的使用者的语音信号和从空气中传播近来的外界环境噪声信号，所述第二声音信号主要为从空气中传播进来的外界环境噪声信号，并且所述第一声音信号和第二声音信号中的外界环境噪声信号具有相关性；
根据所述第一声音信号和第二声音信号确定控制自适应滤波器更新速度的控制参数；
根据所述第二声音信号和所述控制参数对所述第一声音信号进行降噪滤波，输出降噪滤波后的语音信号；
对所述降噪滤波后的语音信号做进一步的降噪和语音高频增强处理。
由上述可见，在本发明的技术方案中，对送话端的语音在声学层面和电子层面分别进行了语音增强。具体来说：在声学层面上，利用具有特定相对位置关系的主振动传声器和辅振动传声器，分别拾取包括使用者的语音信号和外界环境噪声信号的第一语音信号和以外界环境噪声信号为主的第二声音信号，由于采用了振动传声器结构，因此在拾取时就能将外界噪声衰减20～30dB，并且第一声音信号和第二声音信号的外界环境噪声具有高度的相关性，这为电子层面上的语音增强算法提供了较好的噪声参考信号；在电子层面上首先根据第一声音信号和第二声音信号，确定控制自适应滤波器更新速度的控制参数，然后根据第二声音信号和所述控制参数对所述第一声音信号进行降噪滤波得到信噪比较高的语音信号，最后对降噪滤波后的语音信号做进一步的降噪和语音高频增强处理，从而大大改善了送话端语音的可懂度和清晰度。可见通过上述的声学层面和电子层面的语音增强处理，最终能够在通信的送话端提供高达40～50dB的降噪量, 极大地提高通信送话端的语音信噪比，并较好地改善了送话端语音的自然度和可懂度，极大提高了高强度噪声环境下的语音信噪比和语音质量。
附图简要说明
图1为带有胶套的麦克风构成的振动传声器的结构示意图；
图2为根据本发明的语音增强装置中装配在支杆上的主、辅振动传声器的结构示意图；
图3A为主振动传声器与耳机佩戴者头部耦合位置示意图；
图3B为应用本发明的带有传声器支杆的耳机与佩戴者腮部耦合的效果示意图；
图4为本发明中电子层面语音增强的系统方框图；
图5为本发明的语音增强方法的具体流程示意图；
图6为本发明的语音增强装置的方框图；
图7为本发明的头戴式降噪通信耳机的方框图。
在所有附图中相同的标号指示相似或相应的特征或功能。
实施本发明的方式
以下将结合附图对本发明的具体实施例进行详细描述。
本发明的语音增强方法包括两大部分，第一部分是声学层面上进行语音增强，并为电子层面上的语音增强算法提供较好信噪比的主信号和与主信号具有高度相关性的噪声参考信号；第二部分是采用声学信号处理方法，进一步对信号进行语音增强处理，提高语音的信噪比，改善送话端语音的可懂度和舒适度。下面将对声学层面和电子层面上的语音增强技术方案分别进行阐述。
在声学层面上，本发明采用双振动传声器结构，主振动传声器和辅振动传声器具有相似的结构并且在空间位置上相互靠近，即主振动传声器和辅振动传声器具有特定相对位置关系。该特定相对位置关系使得主振动传声器拾取通过耦合振动方式传过来的使用者的语音信号和从空气中传播进来的外界环境噪声信号，而辅振动传声器主要拾取从空气中传播进来的外界环境噪声信号，并且分别从空气中传播进主振动传声器和辅振动传声器的外界环境噪声信号具有相关性。具体来说，主振动传声器与耳机佩戴者直接接触，通过耦合振动的方式有效拾取耳机佩戴者的语音信号，辅振动传声器不与耳机佩戴者直接接触，不耦合通过振动传过来的语音信号。对于空气中传播过来的噪声信号，主、辅振动传声器都会大约20～30dB的衰减，并且通过调整主、辅传声器的位置可以保证两个振动传声器拾取到的噪声信号有比较好的相关性。
在本发明的一个具体实施方式中，采用具有密闭胶套结构的麦克风作为振动传声器。图1为麦克风放在密闭胶套中构成振动传声器的结构示意图，如图1所示，麦克风（MIC）10放置在密闭胶套20中，并且在麦克风10的振膜与胶套20之间保留一定的密闭空气腔30以供声音信号通过。从空气当中传播过来的外界环境噪声因为要经过胶套20的衰减才能被麦克风10的振膜拾取到，所以噪声会被大大降低；而对于耦合在胶套20上表面的振动信号，胶套20表面的振动会直接导致密闭空气腔30体积的变化，从而引起麦克风10振膜的振动，所以胶套20上表面的振动信号会被麦克风10有效的拾取到。
另外，带有胶套20的麦克风10在隔离外界噪声的同时必须有效耦合耳机佩戴者的语音信号，一般人讲话时，人头部的很多部分都会包含一定的语音振动信号（尤其是低频信息），而这其中又以喉部和腮部振动包含的语音频谱信息较丰富。因此，考虑到耳机的佩戴方便和美观，在本发明的一个优选实施方式中，设计如图2所示的传声器支杆，支杆头部的正反两面各放置一个带有胶套的麦克风，分别称为主振动传声器112和辅振动传声器114，其中主振动传声器112设置在贴近佩戴者面部的一面，辅振动传声器114设置在与主振动传声器112相背的另一面。主振动传声器112与耳机佩戴者头部的耦合位置可以有多种选择，图3A示出了主振动传声器与头部耦合的可能位置示意图，包括头顶301、额部302、腮部303、鬓部304、耳内305、耳后306、喉部307等，带有传声器支杆的耳机与佩戴者腮部耦合效果如图3B所示。主振动传声器112的胶套正面与耳机佩戴者的腮部保持较好的耦合，从而能较好拾取耳机佩戴者的语音信息。而辅振动传声器114不与人脸直接耦合，所以对耳机佩戴者语音信号不敏感。
并且，采用如图1所示的胶套结构和如图2、图3B所示的支杆和耳机佩戴方式，能够保证主振动传声器112拾取到的是较好的语音信号和被衰减大约20～30dB的外界噪声信号，辅振动传声器114拾取到的主要是被衰减大约20～30dB的外界噪声信号，辅振动传声器114拾取到的较纯净的外界噪声信号可以为下一步电子层面的降噪提供较好的外界噪声参考信号。在空间上主振动传声器112、辅振动传声器114距离相对较近，并有相似的胶套结构，保证泄露进两个胶套的外界噪声信号具有较好的相关性，以确保电子层面对噪声信号能够进一步降低。
另外为了避免辅振动传声器114拾取到较多的振动语音信号，从而导致在电子层面损伤主振动传声器112中的语音信号，最好能够在主振动传声器112、辅振动传声器114之间采取较好的隔振处理措施。在本发明的一个优选实施方式中，采用在主、辅传声器胶套之间增加一些垫片来达到隔振的目的。
经过声学层面的语音增强后，主振动传声器112中信号的信噪比有了大约20dB提高，但还是不能满足在极限噪声情况下通信的要求。所以在本发明中，采用声学信号处理的技术进一步提高语音信号的信噪比，并改善通过振动拾取的语音信号的自然度和清晰度。
需要说明的是，本发明中的振动传声器并不仅限于上述具有密闭胶套结构的麦克风，也可以采用现有的骨导传声器，或者采用普通的驻极体（ECM）麦克风增加特殊声学结构设计来达到类振动传声器的效果。本发明后续会针对采用普通麦克风加特殊声学结构设计来阐述。
图4为对经过声学层面语音增强后的信号进行电子层面语音增强的系统方框图。如图4所示，电子层面的语音增强，主要包括语音检测模块210、自适应滤波模块220和后处理模块230，其中语音检测模块210用于根据主振动传声器112和辅振动传声器114输出的声音信号确定自适应滤波模块220的更新速度并输出控制参数α；自适应滤波模块220则根据辅振动传声器114输出的声音信号和语音检测模块210输出的控制参数α对主振动传声器112输出的声音信号进行降噪滤波，并输出降噪后的语音信号；后处理模块230用于对采用自适应滤波模块220输出的降噪滤波后的语音信号做进一步的降噪和语音高频增强处理。
当存在语音信号时，主振动传声器112直接耦合佩戴者腮部的振动拾取到较大的语音信号；虽然辅振动传声器114不与腮部直接耦合，但因为其与佩戴者嘴部距离较近，当佩戴者大声讲话时，通过空气泄露而被辅振动传声器114拾取到的语音信号也不能够被忽略。这时如果把辅振动传声器114的信号直接作为滤波参考信号来更新自适应滤波器并进行滤波，将有可能对语音造成损伤，所以必须先由语音检测模块210根据主振动传声器112和辅振动传声器114输出的声音信号确定自适应滤波模块220中自适应滤波器的更新速度，并输出表示控制自适应滤波器221更新速度的控制参数α。
在本发明的一个具体实施方式中，控制参数α的值是采用计算在低频范围内主振动传声器112与辅振动传声器114的统计能量比值P_ratio来确定的，能量比值P_ratio越大表示主振动传声器112所拾取的声音信号中存在目标语音的比例越大，α的值就越小，自适应滤波器的更新速度就越慢；反之，能量比值P_ratio越小则表示表示主振动传声器112所拾取的声音信号中存在目标语音的比例越小、存在环境噪声的比例越大，α的值就越大，自适应滤波器221的更新速度就越快。低频范围是指500Hz以下的频率范围。α的取值范围为0≤α≤1，在本发明的一个优选实施方式中，设定P_ratio大于10dB时，认为主振动传声器112所拾取的声音信号全部为目标语音信号，α=0，自适应滤波器停止更新；P_ratio小于0dB时，认为主振动传声器112所拾取的声音信号全部为环境噪声信号，α=1，自适应滤波器以最快速度更新。
自适应滤波模块220包括一个自适应滤波器221和一个减法器222，在本发明的一个具体实施方式中，采用一个阶长为P（P≥1）的FIR滤波器作为降噪滤波的自适应滤波器，滤波器的权值是 ，本实施方案P=64，阶长主要取决于系统采样频率和主、辅传声器之间声学传递路径的复杂性。
假设主振动传声器112和辅振动传声器114所拾取并输出的声音信号分别为第一声音信号s1(n)和第二声音信号s2(n)，自适应滤波器221的输入信号为辅振动传声器114所拾取的声音信号s2(n)，在控制参数α的更新速度控制下，自适应滤波器221滤波输出信号s3(n)，减法器222将s3(n)与主振动传声器112所拾取的声音信号s1(n)相减得到噪声抵消后的信号y(n)，y(n)反馈回自适应滤波器221进行滤波器权值的再次更新。
自适应滤波器221的更新速度受控制参数α的控制，当α=1，即s1(n)、s2(n)中全是噪声成分，自适应滤波器221快速收敛到噪声从辅振动传声器114到主振动传声器112的传递函数H_noise，使得s3(n)与s1(n)相同，抵消后的y(n)很小，从而消除噪声。当α=0，即s1(n)、s2(n)中全是目标语音成分，自适应滤波器停止更新，从而自适应滤波器不会收敛到语音从辅振动传声器114到主振动传声器112的传递函数H_speech，s3(n)与s1(n)不同，从而相减后的语音成分不会被抵消，输出y(n)保留了语音成分。当0<α<1时，即主振动传声器112所拾取到的声音信号中同时有语音成分和环境噪声成分，这时自适应滤波器221的更新速度由语音成分和环境噪声成分的多少来控制，以保证消除噪声的同时保留语音成分。
另外，由于噪声从辅振动传声器114到主振动传声器112的传递函数H_noise与语音从辅传声器114到主振动传声器112的传递函数H_speech有相似性，因此即使自适应滤波器221收敛到H_noise仍然会对语音造成一定程度的损害，因此需要采用α来约束自适应滤波器221的权值。在本发明的一个具体实施方式中所做的约束是 ，当α=1，即认为主振动传声器112所拾取到的声音信号中全是环境噪声成分，自适应滤波器221不做约束，环境噪声被完全消除；当α=0，即认为主振动传声器112所拾取到的声音信号中全是语音成分，自适应滤波器221完全约束，语音完全保留；当0<α<1时，即认为主振动传声器112所拾取到的声音信号中同时有语音成分和环境噪声成分，自适应滤波器221部分约束，环境噪声部分消除而将语音完全保留，通过这种处理方式达到在降噪的同时很好地保护语音的效果。
需要说明的是，虽然在上述具体实施方式中是利用时域自适应滤波器进行降噪，但本领域技术人员应当明了，在滤波时所采用的滤波器并不限于时域自适应滤波器，也可利用频域（子带）自适应滤波器降噪，进一步可通过主振动传声器112与辅振动传声器114的每个频率子带的统计能量比P_ratioi得到每个频率子带的控制参数αi，并独立控制频率自适应滤波器每个频率子带的更新。i为频率子带的标识，其中每个频率子带的统计能量比越大，该频率子带对应的αi的值越小，αi的取值范围为0≤αi≤1，即αi的取指范围为0到1。
在本发明的一个优选实施方式中，后处理模块230包括单通道降噪子模块231和语音高频增强子模块232。单通道降噪子模块231首先根据噪声平稳的特性统计出自适应滤波模块220的输出信号y(n)中残留的平稳噪声的能量；另外，由于振动方式拾取到的语音信号高频能量较小，导致处理后的语音的清晰度和可懂度不高，因此再采用语音高频增强子模块232对经过单通道降噪子模块231做单通道降噪处理后的语音信号进行高频成分的增强，从而极大的改善输出语音信号的清晰度和可懂度，使得用户获得足够清晰的语音信号。
在本发明的一个具体实施方式中，单通道降噪子模块231利用平滑平均的方法统计出噪声能量，并在信号y(n)中减去这部分噪声能量，从而进一步降低自适应滤波模块220所输出的y(n)中的噪声成分并保留其中的语音成分，以达到提高语音信号信噪比的效果。
结合上述对本发明的技术方案的表述，图5为本发明提供的语音增强方法的具体流程示意图。如图5所示，本发明的语音增强方法包括如下步骤：
首先，在步骤S510中，利用主振动传声器112和辅振动传声器114分别拾取第一声音信号s1(n)和第二声音信号s2(n)，其中第一声音信号s1(n)包括通过耦合振动方式传过来的使用者的语音信号和从胶套泄露进麦克风的外界环境噪声信号，第二声音信号s2(n)主要为从胶套泄露进麦克风的外界环境噪声信号，并且由于振动传声器的位置设置使得第一声音信号s1(n)和第二声音信号s2(n)中的外界环境噪声信号具有相关性；
在步骤S520中，根据第一声音信号s1(n)和第二声音信号s2(n)确定自适应滤波器的更新速度并输出控制参数α，0≤α≤1；
在步骤S530中，根据第一声音信号s1(n)、第二声音信号s2(n)和所述控制参数α利用自适应滤波器对第一声音信号s1(n)进行降噪处理；
在S540中，进一步消除自适应滤波器进行降噪处理后的声音信号中残留的平稳噪声的能量；
最后，在步骤S550中，对上述消除残留的平稳噪声的能量后的声音信号进行高频成分的增强。
本发明的上述语音增强方法，采用软件和硬件组合的方式实现。
图6示出了本发明的与上述语音增强方法相对应的语音增强装置的逻辑结构示意图。如图6所示，本发明提供的语音增强装置600包括声学语音增强单元610和电子语音增强单元620。
其中，声学语音增强单元610包括主振动传声器112和辅振动传声器114。主振动传声器112用于拾取通过耦合振动方式传过来的使用者的语音信号和从空气中传播进来的外界环境噪声信号；辅振动传声器114用于拾取从空气中传播进来的外界环境噪声信号；并且分别从空气中传播进主振动传声器112和辅振动传声器114的外界环境噪声信号具有相关性。
电子语音增强单元620包括语音检测模块210、自适应滤波模块220和后处理模块230，其中，语音检测模块210用于根据所述主振动传声器112和辅振动传声器114输出的声音信号确定所述自适应滤波模块220的更新速度并输出控制参数α；自适应滤波模块220根据所述辅振动传声器114输出的声音信号和所述语音检测模块210输出的控制参数α对所述主振动传声器112输出的声音信号进行降噪滤波，并输出降噪滤波后的语音信号；所述后处理模块230用于对所述自适应滤波模块220输出的降噪滤波后的语音信号做进一步的降噪和语音高频增强处理。
这里需要说明的是：
当自适应滤波器221为时域自适应滤波器时：语音检测模块210，用于通过计算在低频范围内的主振动传声器112输出的声音信号和辅振动传声器114输出的声音信号的统计能量比值来确定自适应滤波器221的控制参数；其中统计能量比值越大，所述控制参数的值越小，且所述控制参数的取值范围为0到1；
当自适应滤波器221为频域自适应滤波器时：语音检测模块210，用于通过计算主振动传声器112输出的声音信号和辅振动传声器114输出的声音信号在每个频率子带的统计能量比值来确定每个频率子带的控制参数αi；其中频率子带的统计能量比值越大，该频率子带对应的控制参数αi的值越小，且每个频率子带对应的控制参数αi的取值范围为0到1。
语音增强装置600各组成结构间的具体工作流程与前述对图4以及图5中所表述的工作流程完全相同，在此不再赘述。
图7示出了具有根据本发明的语音增强装置的头戴式降噪通信耳机700的方框图。
如图7所示，所述头戴式降噪通信耳机700包括语音信号传送端口701以及如图6所示的所述语音增强装置600，其中语音信号传送端口701用于把近端语音信号传送到远端用户，即接收采用语音增强装置600降噪后的语音信号，然后采用有线或者是无线的方式发送给远端用户。所述语音增强装置600的各个组成部件的功能及其描述与上面针对图4以及图6进行的描述完全相同，在此不再进行描述。
综合来讲，本发明能够从声学层面和电子层面消除环境噪声，极大提高高强度噪声环境下语音信噪比和语音质量原因如下：
1）双振动传声器能有效隔离外界从空气中传播过来的噪声；并且对于泄露进来的噪声，因为主、辅振动传声器具有相似的结构和相互靠近的空间位置，所以泄露进主、辅振动传声器的外界噪声信号具有很好的相关性。
2）对于耳机佩戴者讲话时的有用的语音信号，因为主振动传声器是直接和人的头部耦合的，而且主、辅振动传声器之间较好隔离，所以主振动传声器能较好拾取耳机佩戴者的振动语音信号，而辅振动传声器只能拾取泄露进来的语音信号。
3）经过声学层面的语音增强，得到较高信噪比的语音信号和较纯净的外界噪声参考信号，在电子层面采用自适应噪声消除技术和单通道语音增强技术进一步提高语音信号的信噪比。
4）在电子层面对经过语音增强后的语音信号进行高频成分的增强，从而极大的改善输出语音信号的清晰度和可懂度，使得用户获得足够清晰的语音信号。
5）与采用近讲麦克风作为送话器的通信耳机相比，本发明对噪声的方向性和所处位置不敏感，对近、远场的各个方向的噪声都有稳定的降噪量，而且对风噪声也有较好的降噪效果。
如上参照附图以示例的方式描述根据本发明的语音增强方法、装置和降噪耳机。但是，本领域技术人员应当理解，对于上述本发明所提出的语音增强方法、装置和降噪耳机，还可以在不脱离本发明内容的基础上做出各种改进。因此，本发明的保护范围应当由所附的权利要求书的内容确定。

Claims (1)

  1. 1、一种语音增强装置，其特征在于，该装置包括：声学语音增强单元和电子语音增强单元；其中，
    声学语音增强单元包括：具有特定相对位置关系的主振动传声器和辅振动传声器；所述特定相对位置关系使得主振动传声器拾取通过耦合振动方式传过来的使用者的语音信号和从空气中传播进来的外界环境噪声信号，辅振动传声器主要拾取从空气中传播进来的外界环境噪声信号，并且主振动传声器和辅振动传声器所拾取的从空气中传播进来的外界环境噪声信号具有相关性；
    电子语音增强单元包括：语音检测模块、自适应滤波模块和后处理模块；其中，
    语音检测模块，用于根据所述主振动传声器和辅振动传声器输出的声音信号确定所述自适应滤波模块的更新速度并输出控制参数；
    自适应滤波模块，用于根据所述辅振动传声器输出的声音信号和所述语音检测模块输出的控制参数对所述主振动传声器输出的声音信号进行降噪滤波，并输出降噪滤波后的语音信号；
    后处理模块，用于对所述自适应滤波模块输出的降噪滤波后的语音信号做进一步的降噪和语音高频增强处理。
    2、根据权利要求1所述的装置，其特征在于，
    所述主振动传声器由麦克风放在密闭的胶套中构成，并且麦克风的振膜与胶套之间设置有密闭空气腔；
    所述辅振动传声器的结构与所述主振动传声器的结构相同。
    3、根据权利要求1所述的装置，其特征在于，
    所述主振动传声器和辅振动传声器分别放置在传声器支杆的正反两面，且主振动传声器和辅振动传声器之间有隔振处理结构。
    4、根据权利要求1所述的装置，其特征在于，所述后处理模块包括：
    单通道降噪子模块，用于统计出自适应滤波模块输出的降噪滤波后的语音信号中残留的平稳噪声的能量，并从自适应滤波模块输出的降噪滤波后的语音信号中减去这部分噪声能量，然后输出给语音高频增强子模块；
    语音高频增强子模块，用于对单通道降噪子模块降噪处理后的语音信号进行高频成分的增强处理。
    5、根据权利要求1所述的装置，其特征在于，
    所述语音检测模块，用于通过计算在低频范围内的主振动传声器输出的声音信号和辅振动传声器输出的声音信号的统计能量比值来确定所述控制参数；其中统计能量比值越大，所述控制参数的值越小，且所述控制参数的取值范围为0到1；
    或者，
    所述语音检测模块，用于通过计算主振动传声器输出的声音信号和辅振动传声器输出的声音信号在每个频率子带的统计能量比值来确定每个频率子带的控制参数；其中频率子带的统计能量比值越大，该频率子带对应的控制参数的值越小，且每个频率子带对应的控制参数的取值范围为0到1。
    6、根据权利要求1所述的装置，其特征在于，所述自适应滤波模块包括：自适应滤波器和减法器； 其中，
    自适应滤波器，用于在所述控制参数的控制下对辅振动传声器输出的声音信号进行滤波，并输出给减法器；
    减法器，用于将主振动传声器输出的声音信号与自适应滤波器输出的信号相减后输出降噪滤波后的语音信号，并且将该降噪滤波后的语音信号反馈给自适应滤波器。
    7、一种头戴式降噪通信耳机，其特征在于，该通信耳机包括语音信号传送端口和如权利要求1-6中任一项所述的语音增强装置；
    所述语音信号传送端口，用于接收所述语音增强装置降噪后的语音信号，并传送给远端用户。
    8、一种语音增强方法，其特征在于，该方法包括：
    利用具有特定相对位置关系的主振动传声器和辅振动传声器，分别拾取第一声音信号和第二声音信号；所述第一声音信号包括通过耦合振动方式传过来的使用者的语音信号和从空气中传播近来的外界环境噪声信号，所述第二声音信号主要为从空气中传播进来的外界环境噪声信号，并且所述第一声音信号和第二声音信号中的外界环境噪声信号具有相关性；
    根据所述第一声音信号和第二声音信号确定控制自适应滤波器更新速度的控制参数；
    根据所述第二声音信号和所述控制参数对所述第一声音信号进行降噪滤波，输出降噪滤波后的语音信号；
    对所述降噪滤波后的语音信号做进一步的降噪和语音高频增强处理。
    9、根据权利要求8所述的方法，其特征在于，对所述降噪滤波后的语音信号做进一步的降噪和语音高频增强处理包括：
    统计出所述降噪滤波后的语音信号中残留的平稳噪声的能量，并从所述降噪滤波后的语音信号中减去这部分噪声能量，然后再进行高频成分的增强处理。
    10、根据权利要求8或9所述的方法，其特征在于，所述根据所述第一声音信号和第二声音信号确定控制自适应滤波器更新速度的控制参数包括：
    通过计算在低频范围内的第一声音信号和第二声音信号的统计能量比值来确定所述控制参数，其中统计能量比值越大，所述控制参数的值越小，且所述控制参数的取值范围为0到1；
    或者，
    通过计算第一声音信号和第二声音信号的在每个频率子带的统计能量比值来确定每个频率子带的控制参数，其中频率子带的统计能量比值越大，该频率子带对应的控制参数的值越小，且每个频率子带对应的控制参数的取值范围为0到1。
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