WO2016000585A1 - 提高免提通话设备通话质量的方法、装置和免提通话设备 - Google Patents

提高免提通话设备通话质量的方法、装置和免提通话设备 Download PDF

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Publication number
WO2016000585A1
WO2016000585A1 PCT/CN2015/082634 CN2015082634W WO2016000585A1 WO 2016000585 A1 WO2016000585 A1 WO 2016000585A1 CN 2015082634 W CN2015082634 W CN 2015082634W WO 2016000585 A1 WO2016000585 A1 WO 2016000585A1
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WIPO (PCT)
Prior art keywords
angle
voice feature
acquisition
microphone
feature signal
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Ceased
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PCT/CN2015/082634
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English (en)
French (fr)
Inventor
华洋
周宏伟
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Qingdao Goertek Technology Co Ltd
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Qingdao Goertek Technology Co Ltd
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Priority to JP2016548670A priority Critical patent/JP6211716B2/ja
Priority to EP15814169.7A priority patent/EP3089431A4/en
Priority to US15/122,835 priority patent/US9648154B1/en
Priority to KR1020167024556A priority patent/KR101673123B1/ko
Publication of WO2016000585A1 publication Critical patent/WO2016000585A1/zh
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers
    • H04R3/005—Circuits for transducers for combining the signals of two or more microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/60—Substation equipment, e.g. for use by subscribers including speech amplifiers
    • H04M1/6033—Substation equipment, e.g. for use by subscribers including speech amplifiers for providing handsfree use or a loudspeaker mode in telephone sets
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/60—Substation equipment, e.g. for use by subscribers including speech amplifiers
    • H04M1/6033—Substation equipment, e.g. for use by subscribers including speech amplifiers for providing handsfree use or a loudspeaker mode in telephone sets
    • H04M1/6041—Portable telephones adapted for handsfree use
    • G—PHYSICS
    • G01—MEASURING; TESTING
    • G01L—MEASURING FORCE, STRESS, TORQUE, WORK, MECHANICAL POWER, MECHANICAL EFFICIENCY, OR FLUID PRESSURE
    • G01L21/00—Vacuum gauges
    • G01L21/02—Vacuum gauges having a compression chamber in which gas, whose pressure is to be measured, is compressed
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/02—Constructional features of telephone sets
    • H04M1/03—Constructional features of telephone transmitters or receivers, e.g. telephone hand-sets
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724—User interfaces specially adapted for cordless or mobile telephones
    • H04M1/72448—User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/725—Cordless telephones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; ELECTRIC HEARING AIDS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/326—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only for microphones
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Speech or voice signal processing techniques 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
    • G10L2021/02166—Microphone arrays; Beamforming

Definitions

  • the present invention relates to the field of hands-free calling devices, and in particular, to a method, a device and a hands-free calling device for improving the call quality of a hands-free calling device.
  • the invention provides a method, a device and a hands-free calling device for improving the call quality of a hands-free calling device, so as to solve the problem of collecting more environmental noise and low signal-to-noise ratio when the hands-free calling device is in a call.
  • the hands-free calling device includes a calling end composed of a main microphone and at least one auxiliary microphone.
  • the method includes:
  • the transmitter is calibrated to the direction determined by the second acquisition angle.
  • determining a second acquisition angle that is smaller than the initial acquisition angle in the first acquisition angle according to the direction of the voice feature signal includes:
  • the line connecting the main microphone and one of the auxiliary microphones is the axis of symmetry, and the mirror angle ⁇ 2 of the angle ⁇ 1 with respect to the axis of symmetry is defined, and the angle ⁇ 2 is confirmed as the second acquisition angle.
  • whether there is a voice feature signal in the real-time judgment angle ⁇ 1 includes:
  • Envelope energy detection is performed on the voice feature signal in the ⁇ 1 direction, and when the energy detection value is greater than the first preset threshold, the zero-crossing rate detection is performed on the voice feature signal in the ⁇ 1 direction;
  • performing envelope energy detection on the voice feature signal in the ⁇ 1 direction includes:
  • Envelope energy detection is performed by the following formula:
  • power is the energy value of the speech feature signal
  • parameter alpha is a weighting factor
  • parameter N is a speech feature signal at a certain point
  • Zero-crossing detection of the speech feature signal in the ⁇ 1 direction includes: zero-crossing detection by the following formula:
  • Z_rate is the zero-crossing rate of the speech feature signal
  • n is a value in the discrete time series.
  • the hands-free calling device further includes a receiving end composed of at least one speaker, and the method further includes:
  • a virtual speaker of a speaker in the receiving end is set, the connection of the speaker and its virtual speaker is directed to the ⁇ 1 direction, and a third acquisition angle is defined, and the third acquisition angle is directed to the direction determined by the second acquisition angle.
  • an apparatus for improving the call quality of a hands-free calling device includes:
  • a voice feature determining unit configured to scan within a first first acquisition angle of the sending end; after scanning the voice feature signal within the first collecting angle, determining, according to the direction of the voice feature signal, within the first collecting angle a second acquisition angle that is smaller than the first acquisition angle;
  • the direction calibration unit is configured to calibrate the sending end to the direction determined by the second acquisition angle.
  • the device further includes: a virtual microphone creating unit, configured to scan the ⁇ 1 direction with a voice feature signal in the initial first acquisition angle of the sending end, and perform a reverse extension line through the main microphone along the ⁇ 1 direction. Taking the main microphone as the center, the circle is drawn by the main microphone and one of the auxiliary microphone lines, and the position where the arc intersects with the reverse extension line is determined as the virtual microphone of the main microphone;
  • the angle confirmation unit is configured to use the virtual microphone of the main microphone and the main microphone as a new voice array, and define an angle ⁇ 1 smaller than the first acquisition angle, and determine whether there is a voice feature signal in the angle ⁇ 1 in real time; ⁇ 1 is the second acquisition angle; otherwise, the line connecting the main microphone and one of the auxiliary microphones is the axis of symmetry, and the mirror angle ⁇ 2 of the angle ⁇ 1 with respect to the axis of symmetry is defined, and the angle ⁇ 2 is confirmed as the second acquisition angle.
  • the angle confirmation unit is further used,
  • Envelope energy detection is performed on the voice feature signal in the ⁇ 1 direction, and when the energy detection value is greater than the first preset threshold, the zero-crossing rate detection is performed on the voice feature signal in the ⁇ 1 direction;
  • the confirmation angle ⁇ 1 contains a speech feature.
  • the angle confirmation unit comprises:
  • An envelope detecting unit is configured to perform envelope energy detection on the voice feature signal in the ⁇ 1 direction by the following formula:
  • the zero-crossing detection unit is configured to perform zero-crossing rate detection on the voice feature signal in the ⁇ 1 direction by the following formula:
  • Z_rate is the zero-crossing rate of the speech feature signal
  • n is a value in the discrete time series.
  • the device further includes: a receiving end positioning unit, configured to set a virtual speaker of a speaker in the receiving end, the connection of the certain speaker and the virtual speaker thereof is directed to the ⁇ 1 direction, and defining a third collecting angle,
  • the range of regions included in the third acquisition angle includes the direction determined by the second acquisition angle.
  • the method and device for improving the call quality of the hands-free calling device of the present invention reduces the voice pickup angle of the sending end to a relatively accurate range by performing voice calibration positioning on the sending end, thereby avoiding the sending end
  • the picked up voice signal includes more environmental noise, affecting the signal-to-noise ratio of the voice, solves the problem that the call quality of the hands-free calling device is not high, and improves the call quality of the hands-free calling device.
  • FIG. 1 is a flowchart of a method for improving call quality of a hands-free calling device according to an embodiment of the present invention
  • FIG. 2 is a schematic diagram of a method for improving a call quality of a hands-free calling device according to an embodiment of the present invention
  • FIG. 3a is a schematic diagram of a transmitter terminal calibration according to an embodiment of the present invention.
  • FIG. 3b is a schematic diagram of a transmitter terminal calibration according to an embodiment of the present invention.
  • FIG. 4 is a flowchart of determining a voice feature according to an embodiment of the present invention.
  • FIG. 5 is a schematic diagram of calibration of a receiving end according to an embodiment of the present invention.
  • FIG. 6 is a block diagram of an apparatus for improving call quality of a hands-free calling device according to an embodiment of the present invention.
  • the core idea of the present invention is to track the direction of the voice source in real time through the microphone array sound source localization technology, and at the same time determine a voice protection angle smaller than the initial voice protection angle for sound pickup, which can greatly reduce the interference of the surrounding environment noise.
  • the purpose of improving the signal-to-noise ratio is improved.
  • the orientation information of the voice source is used to compensate the direction of the speaker array, so that the sound emitted just points to the voice source, which improves the privacy of the received speech.
  • FIG. 1 is a flowchart of a method for improving call quality of a hands-free calling device according to an embodiment of the present invention
  • the hands-free calling device includes a sending end composed of a main mic and at least one auxiliary mic, the method include:
  • Step S110 performing scanning within the initial first acquisition angle of the sending end
  • Step S120 after scanning the voice feature signal in the first acquisition angle, determining a second acquisition angle smaller than the first acquisition angle in the first acquisition angle according to the direction of the voice feature signal;
  • Step S130 calibrating the sending end to the direction determined by the second collecting angle.
  • the voice protection angle of the sending end is reduced to a relatively small angle, and the sound is picked up within a re-determined small angle, thereby avoiding the interference of the environmental noise on the voice signal, and improving the sent message.
  • the noise ratio which in turn improves the call quality of the hands-free calling device.
  • the hands-free calling device 1 includes: a sending end and a receiving end;
  • the transmitting end comprises a main microphone MIC-a and a auxiliary microphone MIC-b
  • the receiving end comprises a main speaker SPK-a and a auxiliary speaker SPK-b
  • the angle ⁇ is the initial scanning angle of the sending end.
  • the sending end collects the signal, first scans according to the larger collecting angle ⁇ , and after detecting the speech feature in the angle ⁇ , then reduces the collecting angle to the angle.
  • ⁇ achieves the purpose of voice positioning at the sending end.
  • the process of voice location of the sending end is: the sending end scans the voice feature signal within an initial scanning angle ⁇ , and when determining the voice feature in the ⁇ angle, the sending end is positioned to the direction determined by the angle,
  • the direction determined by the angle is the position or orientation of the user's speech (as shown in Fig.
  • the range of the angle ⁇ is open to cover the position of the human head, and the positioning of the human head is more accurate than the angle ⁇ ).
  • the method of the invention has the function of reducing the initial scanning angle to a smaller second acquisition angle ⁇ , so that the speech protection angle is smaller, the position of the user can be more accurately positioned, and the environmental noise in the speech signal is reduced. Improve voice quality.
  • FIG. 3a is a schematic diagram of the terminal calibration according to an embodiment of the present invention. referring to FIG. 3a, reference numeral 1 denotes a hands-free calling device, 2 denotes a secondary speaker, 3 denotes a secondary microphone, and 4 denotes a virtual microphone of the primary microphone.
  • the specific implementation manner of the improved hands-free calling device of the present invention is:
  • the transmitting end when the communication is performed, the transmitting end first performs a voice scanning at a first first acquisition angle ⁇ , and after scanning to the first acquisition angle ⁇ , there is a voice feature in the ⁇ 1 direction, and then passes through the ⁇ 1 of the main microphone MIC-a.
  • the virtual microphone MIC-c is disposed on the reverse extension line of the direction, and the virtual microphone MIC-c is located on the arc with the main microphone MIC-a as the center, the main microphone MIC-a and the auxiliary MIC-b as the radius, and then the main The microphone MIC-a and the virtual microphone MIC-c form a new array, defining that the directivity angle of the new array is very small, the second acquisition angle ⁇ 1, and the angle ⁇ 1 points to a smaller range in the ⁇ 1 direction; the value of ⁇ 1 is different according to the application.
  • the scene is specifically selected.
  • one of the core contents of the method provided by the present invention is: using the virtual microphone MIC-c to judge, the method of determining is to first set the virtual microphone MIC-c in the ⁇ 1 direction in FIG. 3a and form a pointing angle ⁇ 1, The speech feature in the pointing angle of ⁇ 1 is judged in real time, and if the speech feature is found, the pointing angle ⁇ 1 is determined as the number of the transmitting end.
  • the transmitting end is calibrated to the direction determined by the second acquisition angle ( ⁇ 1). If the voice feature is not found, then the virtual microphone MIC-c is set again to determine the direction of the voice on the mirror opposite side of the main microphone MIC-a and the auxiliary MIC-b connection.
  • FIG. 3b is a schematic diagram of the terminal calibration according to an embodiment of the present invention. referring to FIG. 3b, reference numeral 1 denotes a hands-free calling device, 2 denotes a secondary speaker, 3 denotes a secondary microphone, and 4 denotes a virtual microphone of the primary microphone.
  • the symmetric angle ⁇ 2 of ⁇ 1 is set on the other side of the line connecting the main microphone MIC-a and the auxiliary microphone MIC-b in Fig. 3b, when determining ⁇ 2 If there is a voice feature within the angle, the transmitter is calibrated to the direction determined by ⁇ 2. After the above steps, the positioning and calibration of the sending end are completed, and the directivity of the pickup is improved.
  • the positioning of the sending end is specifically described below in conjunction with the judgment of the voice feature.
  • FIG. 4 is a flowchart of determining a voice feature according to an embodiment of the present invention. Referring to FIG. 4, whether a voice feature is included in a specific real-time judgment angle ⁇ 1 includes:
  • performing envelope energy detection on the voice feature signal in the ⁇ 1 direction includes:
  • Envelope energy detection is performed by the following formula:
  • parameter alpha is the weighting factor
  • parameter N is the speech feature signal at a certain point; wherein the alpha and N parameters are adjusted to control the detection sensitivity to ensure the envelope energy detection. Stability, once the envelope energy is found to be greater than the first preset threshold (Set according to the actual situation), enter the zero-crossing rate detection step.
  • Zero-crossing detection of the speech feature signal in the ⁇ 1 direction includes: zero-crossing detection by the following formula:
  • Z_rate is the zero-crossing rate of the speech feature
  • n is a value in the discrete time series.
  • the zero-crossing rate Z rate is greater than the second preset threshold, it is considered that the acquired signal within the ⁇ 1 angle has a voice feature signal, and the voice action is determined.
  • the position of the speech end is calibrated and calibrated to the range determined by the angle ⁇ 1. If there is no speech feature signal in the ⁇ 1 angle ( ⁇ 1 is not the direction of the speech source), then it is judged whether there is a speech feature signal in the angle ⁇ 2 which is symmetric with the line of ⁇ 1 with respect to the MIC-a MIC-b, and the envelope energy can also be used at this time. Detection and zero crossing detection further verify the speech signature signal within ⁇ 2.
  • the precise positioning of the transmitting end is realized by virtual array element positioning and voice feature signal detection.
  • FIG. 5 is a schematic diagram of a receiving end calibration according to an embodiment of the present invention.
  • reference numeral 1 denotes a hands-free calling device
  • 2 denotes a secondary speaker
  • 3 denotes a primary microphone.
  • the orientation of the voice source is used to compensate the direction of the speaker array, so that the sound emitted is directed to the voice, that is, the sounding direction of the speaker array is adjusted as much as possible to the position of the user positioned by the sending end. , that is, the direction determined by the second acquisition angle to improve the privacy of the received speech.
  • the virtual array element technology is applied, and the speaker SPK-b is virtualized, and the connection between the virtual speaker SPK-c and the speaker SPK-a is directed to the voice direction, and then a pointing angle is set, and the coverage area of the third acquisition angle is
  • the range of the sound propagation region included in the third acquisition angle covers the direction of the voice determined by the second acquisition angle (as shown in FIG. 5, the angle opening range under the SPK-a covers the position of the human head), preferably
  • the third acquisition angle directly points to the determined speech direction of the second acquisition angle.
  • the hands-free calling device further includes a receiving end composed of at least one speaker, and is determined.
  • the method further comprises: calibrating the receiving end of the hands-free device, setting a virtual speaker of a speaker in the receiving end, and connecting the speaker and the virtual speaker thereof ⁇ 1 direction, and defining a third acquisition angle, the third acquisition angle is directed to the direction determined by the second acquisition angle, and the third acquisition angle is actually an angular range of the played sound, the angle range including the second acquisition angle and direction,
  • the speaker is played as accurately as possible toward the direction of the voice source, narrowing the range of the played voice, and improving the privacy of the received message.
  • the direction of the voice feature determined at the time of the terminal calibration can be used to calibrate the receiver.
  • the virtual speaker SPK-c of the auxiliary speaker SPK-b is disposed, and the virtual speaker SPK-c is disposed on the reverse extension line passing through the main speaker SPK-a and the ⁇ 1 direction, and the virtual speaker SPK-c is in the main speaker SPK -a is the arc of the center, the main speaker SPK-a and the auxiliary speaker SPK-b are connected in a radius, and the main speaker SPK-a and the virtual speaker SPK-c are combined into a new array, and the directivity angle of the new array is defined as The third acquisition angle.
  • the third acquisition angle is directed to the direction determined by the second acquisition angle determined when the transmitter is calibrated, and the terminal positioning calibration is completed.
  • the method for improving the call quality of the hands-free device of the present invention reduces the interference of surrounding environment noise, improves the signal-to-noise ratio of the transmitted call, and improves the hands-free calling device after the calibration of the sending end and the calibration of the receiving end.
  • the purpose of the call quality The direction of the speaker array is compensated, so that the sound is just directed to the voice, which improves the privacy of the received message and enhances the user experience.
  • the method can be applied to a smart device such as a smart watch with a hands-free calling function, which can greatly improve the calling performance of the smart device.
  • FIG. 6 is a block diagram of an apparatus for improving the call quality of a hands-free calling device according to an embodiment of the present invention.
  • the device 600 includes :
  • the voice feature determining unit 601 is configured to perform scanning in the first first acquisition angle of the sending end; after scanning the voice feature signal in the first collecting angle, according to the direction of the voice feature signal, within the first collecting angle Determining a second acquisition angle that is smaller than the first acquisition angle;
  • the direction calibration unit 602 is configured to calibrate the sending end to the direction determined by the second acquisition angle.
  • the device further includes: a virtual microphone creating unit, configured to perform a reverse extension line through the main microphone along the ⁇ 1 direction after scanning the voice feature in the ⁇ 1 direction within the initial first acquisition angle of the sending end, and Taking the main microphone as the center, the circle is drawn by the main microphone and one of the auxiliary microphone lines, and the position where the arc intersects with the reverse extension line is determined as the virtual microphone of the main microphone;
  • a virtual microphone creating unit configured to perform a reverse extension line through the main microphone along the ⁇ 1 direction after scanning the voice feature in the ⁇ 1 direction within the initial first acquisition angle of the sending end, and Taking the main microphone as the center, the circle is drawn by the main microphone and one of the auxiliary microphone lines, and the position where the arc intersects with the reverse extension line is determined as the virtual microphone of the main microphone;
  • the angle confirmation unit is configured to use the virtual microphone of the main microphone and the main microphone as a new voice array, and define a angle ⁇ 1 smaller than the first acquisition angle, and determine whether there is a voice feature signal in the angle ⁇ 1 in real time; ⁇ 1 is the second acquisition angle; otherwise, the line connecting the main microphone and one of the auxiliary microphones is the axis of symmetry, and the mirror angle ⁇ 2 of the angle ⁇ 1 with respect to the axis of symmetry is defined, and the angle ⁇ 2 is confirmed as the second acquisition angle.
  • the angle confirmation unit is further used,
  • Envelope energy detection is performed on the voice feature signal in the ⁇ 1 direction, and when the energy detection value is greater than the first preset threshold, the zero-crossing rate detection is performed on the voice feature signal in the ⁇ 1 direction;
  • the confirmation angle ⁇ 1 contains the voice feature signal.
  • the angle confirmation unit comprises:
  • An envelope detecting unit is configured to perform envelope energy detection on the voice feature signal in the ⁇ 1 direction by the following formula:
  • the zero-crossing detection unit is configured to perform zero-crossing rate detection on the voice feature signal in the ⁇ 1 direction by the following formula:
  • Z_rate is the zero-crossing rate of the speech feature signal
  • n is a value in the discrete time series.
  • the device further comprises: a receiving end positioning unit, configured to set a virtual speaker of a speaker in the receiving end, the connecting line of the speaker and the virtual speaker thereof is directed to the ⁇ 1 direction, and defines a third collecting angle, the third The range of regions included in the acquisition angle includes the direction determined by the second acquisition angle.
  • a receiving end positioning unit configured to set a virtual speaker of a speaker in the receiving end, the connecting line of the speaker and the virtual speaker thereof is directed to the ⁇ 1 direction, and defines a third collecting angle, the third The range of regions included in the acquisition angle includes the direction determined by the second acquisition angle.
  • a hands-free calling device including a transmitting end composed of a main mic and at least one auxiliary mic, a receiving end composed of at least one speaker, and the above-mentioned improving call quality of the hands-free calling device. s installation.
  • the hands-free calling device of the present invention since the hands-free calling device of the present invention has a sending end and a receiving end, the call receiving end and the transmitting end can be improved according to the method for improving the quality of the hands-free calling, but in some applications,
  • the hands-free calling device only has a sending end or a receiving end.
  • the method for improving the quality of the hands-free call at the receiving end according to the specific embodiment of the present invention can improve the receiving end or improve the hands-free calling at the sending end. Quality methods to improve the sender. That is, the method for improving the quality of the hands-free call of the receiving end and the sending end of the present invention can be separately implemented, and details are not described herein again.

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Abstract

本发明公开了一种提高免提通话设备通话质量的方法、装置和免提通话设备，该免提通话设备包括一个主麦克和至少一个辅麦克组成的送话端，该方法包括：在送话端初始的第一采集角度内进行扫描；当在第一采集角度内扫描到语音特征信号后，根据该语音特征信号的方向，在第一采集角度内确定一个比第一采集角度小的第二采集角度；将送话端校准至该第二采集角度确定的方向。本方法使用较小的语音保护角度进行拾音，可以极大地降低周围环境噪声的干扰，达到提高送话信噪比的目的，使得免提通话设备通话时的语音更有指向性，提高通话的质量。

Description

提高免提通话设备通话质量的方法、装置和免提通话设备 技术领域
本发明涉及免提通话设备领域，具体涉及一种提高免提通话设备通话质量的方法、装置和一种免提通话设备。
发明背景
现有技术中的免提通话设备，例如智能手表、智能手环等智能腕戴设备，在进行免提通话模式时，由于智能手表和用户嘴部相对位置的不确定性，通常采用较大的语音保护角度，这样在拾取语音的同时也采集到较多的环境噪声，影响送话的信噪比；同时由于智能手表等免提通话设备扬声器发出的声音不但能被通话人自己听到，也会被其附近的其他人听到，容易泄露比较私密的信息。总之，送话端和受话端的上述缺陷导致了现有技术中免提通话设备的通话质量不高。
发明内容
本发明提供了一种提高免提通话设备通话质量的方法、装置和一种免提通话设备，以解决免提通话设备通话时，采集较多环境噪声，送话信噪比低的问题。
根据本发明的一个方面，提供了一种提高免提通话设备通话质量的方法，该免提通话设备包括一个主麦克和至少一个辅麦克组成的送话端，该方法包括：
在送话端初始的第一采集角度内进行扫描；
当在第一采集角度内扫描到语音特征信号后，根据该语音特征信号的方向，在第一采集角度内确定一个比第一采集角度小的第二采集角度；
将送话端校准至该第二采集角度确定的方向。
其中，在第一采集角度内扫描到语音特征信号后，根据该语音特征信号的方向，在第一采集角度内确定一个比初始采集角度小的第二采集角度包括：
在送话端初始的第一采集角度内扫描到γ1方向有语音特征信号后，沿着γ1方向，经过主麦克做一条反向延长线，并以主麦克为圆心，以主麦克和其中一个辅麦克连线为半径画圆，圆弧与反向延长线相交的位置确定为主麦克的虚拟麦克风；
以主麦克和主麦克的虚拟麦克风为新的语音阵列，并定义一个比第一采集角度小的角度β1，实时判断角度β1内是否有语音特征信号；是则，确认角度β1为第二采集角度；
否则，以主麦克和其中一个辅麦克的连线为对称轴，定义角度β1关于对称轴的镜像角度β2，确认角度β2为第二采集角度。
其中，实时判断角度β1内是否有语音特征信号包括：
对γ1方向的语音特征信号进行包络能量检测，当能量检测值大于第一预设阀值时，对γ1方向的语音特征信号进行过零率检测；
当γ1方向的语音特征的过零率达到第二预设阀值时，确认角度β1内有语音特征信号。
其中，对γ1方向的语音特征信号进行包络能量检测包括：
通过如下公式进行包络能量检测：
Figure PCTCN2015082634-appb-000001
其中，power为语音特征信号的能量值，参数alpha为计权因子，参数N为某一时点上的语音特征信号；
对γ1方向的语音特征信号进行过零率检测包括：通过如下公式进行过零检测：
Figure PCTCN2015082634-appb-000002
其中，Z_rate为语音特征信号的过零率，n为离散时间序列中的一个值，
Figure PCTCN2015082634-appb-000003
其中，免提通话设备还包括至少一个扬声器组成的受话端，该方法还包括：
设置受话端中一个扬声器的虚拟扬声器，该扬声器和其虚拟扬声器的连线指向γ1方向，并定义第三采集角度，第三采集角度指向第二采集角度确定的方向。
根据本发明的另一个方面，提供了一种提高免提通话设备通话质量的装置；该装置包括：
语音特征确定单元，用于在送话端初始的第一采集角度内进行扫描；当在第一采集角度内扫描到语音特征信号后，根据该语音特征信号的方向，在第一采集角度内确定一个比第一采集角度小的第二采集角度；
方向校准单元，用于将送话端校准至该第二采集角度确定的方向。
其中，该装置还包括：虚拟麦克风创建单元，用于在送话端初始的第一采集角度内扫描到γ1方向有语音特征信号后，沿着γ1方向，经过主麦克做一条反向延长线，并以主麦克为圆心，以主麦克和其中一个辅麦克连线为半径画圆，圆弧与反向延长线相交的位置确定为主麦克的虚拟麦克风；
角度确认单元，用于以主麦克和主麦克的虚拟麦克风为新的语音阵列，并定义一个比第一采集角度小的角度β1，实时判断角度β1内是否有语音特征信号；是则，确认角度β1为第二采集角度；否则，以主麦克和其中一个辅麦克的连线为对称轴，定义角度β1关于对称轴的镜像角度β2，确认角度β2为第二采集角度。
其中，角度确认单元还用于，
对γ1方向的语音特征信号进行包络能量检测，当能量检测值大于第一预设阀值时，对γ1方向的语音特征信号进行过零率检测；
当γ1方向的语音特征信号的过零率达到第二预设阀值时，确认角度β1内包含语音特征。
其中，角度确认单元包括：
包络检测单元，用于通过如下公式对γ1方向的语音特征信号进行包络能量检测：
Figure PCTCN2015082634-appb-000004
其中，power为语音特征信号的能量值，参数alpha为计权因子，N为语音特征的具体的值；
过零检测单元，用于通过如下公式对γ1方向的语音特征信号进行过零率检测：
Figure PCTCN2015082634-appb-000005
其中，Z_rate为语音特征信号的过零率，n为离散时间序列中的一个值，
Figure PCTCN2015082634-appb-000006
其中，该装置还包括：受话端定位单元，用于设置受话端中一个扬声器的虚拟扬声器，该某一扬声器和其虚拟扬声器的连线指向γ1方向，并定义第三采集角度，所述第三采集角度所包括的区域范围包含所述第二采集角度确定的方向。
本发明的这种提高免提通话设备通话质量的方法和装置，通过对送话端进行语音校准定位，将送话端的语音拾取角度缩小到一个相对精确的范围内，以此避免了送话端拾取的语音信号中包括较多环境噪声，影响语音的信噪比，解决了免提通话设备通话质量不高的问题，提高免提通话设备的通话质量。
本申请的其它特征和优点将在随后的说明书中阐述，并且，部分地从说明书中变得显而易见，或者通过实施本申请而了解。本申请的目的和其他优点可通过在所写的说明书、权利要求书、以及附图中所特别指出的结构来实现和获得。
附图简要说明
附图用来提供对本发明的进一步理解，并且构成说明书的一部分，与本发明实施例一起用于解释本发明，并不构成对本发明的限制。在附图中：
图1是本发明一个实施例提供一种提高免提通话设备通话质量的方法的流程图；
图2是本发明一个实施例提供的一种提高免提通话设备通话质量的方法的原理示意图；
图3a是本发明一个实施例提供的送话端校准的示意图；
图3b是本发明一个实施例提供的送话端校准的示意图；
图4是本发明一个实施例提供的确定语音特征的流程图；
图5是本发明一个实施例提供的受话端校准示意图；
图6是本发明一个实施例提供的一种提高免提通话设备通话质量的装置的框图。
具体实施方式
本发明的核心思想是通过麦克风阵列声源定位技术来实时的跟踪语音源的方向，同时确定一个比初始语音保护角度小的语音保护角度进行拾音，可以极大地降低周围环境噪声的干扰，达到提高送话信噪比的目的；同时利用语音源的方位信息，对扬声器阵列进行方向补偿，使其发出的声音刚好指向语音源，提高了受话的私密性。
图1是本发明一个实施例提供一种提高免提通话设备通话质量的方法的流程图；参见图1，该免提通话设备包括一个主麦克和至少一个辅麦克组成的送话端，该方法包括：
步骤S110，在送话端初始的第一采集角度内进行扫描；
步骤S120，当在第一采集角度内扫描到语音特征信号后，根据该语音特征信号的方向，在第一采集角度内确定一个比第一采集角度小的第二采集角度；
步骤S130，将送话端校准至该第二采集角度确定的方向。
通过上述步骤，将送话端的语音保护角度缩小到一个比较小的角度内，并在重新确定的较小的角度内进行拾音，因而避免了环境噪声对语音信号的干扰，提高送话的信噪比，进而提高了免提通话设备的通话质量。
图2是本发明一个实施例提供的一种提高免提通话设备通话质量的方法的原理示意图；参见图2，该免提通话设备1包括：送话端和受话端；
送话端包括主麦克MIC-a和辅麦克MIC-b，受话端包括主扬声器SPK-a和辅扬声器SPK-b；
角度α为送话端初始的扫描角度，正常进行通讯时，送话端采集信号，先按照较大的采集角度α进行扫描，在角度α内检测到语音特征后，再减小采集角度到角度β，进而实现送话端语音定位的目的。送话端语音定位的过程为：送话端在一个初始的扫描角度α内进行语音特征信号的扫描，当确定β角度内具有语音特征时，将送话端定位到该角度确定的方向，该角度确定的方向即是用户说话的位置或者方位(如图2中角度β张开的范围恰好地覆盖了人头的位置，相比角度α对人头的定位更准确)。本发明的方法的作用在于将初始的扫描角度缩小到一个较小的第二采集角度β，这样，语音保护角度较小，能够更加精确的定位到用户说话的位置，减少语音信号中的环境噪声，提高语音质量。
图3a是本发明一个实施例提供的送话端校准的示意图；参见图3a，附图标记1表示免提通话设备、2表示辅扬声器、3表示辅麦克、4表示主麦克的虚拟麦克风。
本发明的提高免提通话设备的具体实现方式为：
参见图3a，进行通讯时，送话端先在较大的第一采集角度α进行语音扫描，当扫描到第一采集角度α内γ1方向有语音特征后，再通过主麦克MIC-a的γ1方向的反向延长线上设置虚拟麦克风MIC-c，虚拟麦克风MIC-c位于以主麦克MIC-a为圆心、主麦克MIC-a和辅助MIC-b连线为半径的弧形上，之后主麦克MIC-a和虚拟麦克风MIC-c组成新的阵列，定义新阵列的指向性角度为非常小的第二采集角度β1，角度β1指向γ1方向的较小范围内；β1的值根据不同的应用场景进行具体的选择。
由于缺少辅助阵元，两个点阵元(MIC-a和MIC-c)组成的阵列，难以区分语音方向来自主麦克MIC-a和辅助MIC-b连线的左侧(γ1)还是右侧(γ2)，本发明提供的方法的核心内容之一就是：利用虚拟麦克风MIC-c来判断，判断的方法是，首先在图3a中的γ1方向设置虚拟麦克风MIC-c并形成指向角β1，实时地判断β1指向角内的语音特征，如果发现了语音特征就将指向角β1确定为送话端的第 二采集角度，将送话端校准至该第二采集角度(β1)确定的方向。如果没有发现语音特征，那么立刻重新在主麦克MIC-a和辅助MIC-b连线的镜像异侧设置虚拟麦克风MIC-c判断语音的方向。
图3b是本发明一个实施例提供的送话端校准的示意图；参见图3b，附图标记1表示免提通话设备、2表示辅扬声器、3表示辅麦克、4表示主麦克的虚拟麦克风。
若在图3a所述β1的指向角度内没有发现语音特征，则在图3b所述主麦克MIC-a和辅麦克MIC-b的连线的另一侧设置β1的对称角β2，当确定β2的角度内有语音特征，则将送话端校准至β2确定的方向。经过上述步骤完成送话端的定位和校准，提高拾音的指向性。
下面结合语音特征的判断来具体说明送话端的定位。
图4是本发明一个实施例提供的确定语音特征的流程图，参见图4，具体的实时判断角度β1内是否有语音特征包括：
1、采集信号，其中，该采集的信号即扫描到的γ1方向的语音特征；
2、将该语音特征进行包络检测，并判断能量值是否大于预设的第一阀值，是则，将该语音特征进行过零率检测；否则，返回重新确定语音特征的方向和语音特征信号。
3、对该语音特征信号的过零率进行判断，当该过零率大于第二预设阀值时，确定该采集角度β1内具有语音特征信号；将该指向角β1作为较小的第二采集角度进而根据该第二采集角度对送话端进行校准。
在本实施例中，对γ1方向的语音特征信号进行包络能量检测包括：
通过如下公式进行包络能量检测：
Figure PCTCN2015082634-appb-000007
其中，power为语音特征信号的能量值，参数alpha为计权因子，参数N为某一时点上的语音特征信号；其中通过alpha和N两个参数调节来控制检测灵敏度，保证包络能量检测的稳定性，一旦发现包络能量power大于第一预设阀值 (根据实际情况设定)，进入过零率检测步骤。
对γ1方向的语音特征信号进行过零率检测包括：通过如下公式进行过零检测：
Figure PCTCN2015082634-appb-000008
其中，Z_rate为语音特征的过零率，n为离散时间序列中的一个值，
Figure PCTCN2015082634-appb-000009
当过零率Z rate大于第二预设阈值时，认为β1角度内的采集信号有语音特征信号，判定语音动作。
在判断β1角度内具有语音特征信号后，将送话端的定位和校准至角度β1确定的范围。如果β1角度内没有语音特征信号(γ1不是语音源方向)，那么判断在与β1关于MIC-a MIC-b的连线对称的角度β2内是否有语音特征信号，此时也可以运用包络能量检测和过零检测进一步验证β2内的语音特征信号。
综上，通过虚拟阵元定位及语音特征信号检测实现了送话端的精确定位。
图5是本发明一个实施例提供的受话端校准示意图；附图标记1表示免提通话设备、2表示辅扬声器、3表示主麦克。
在送话端定位完成后，利用语音源的方位信息，对扬声器阵列进行方向补偿，使其发出的声音刚好指向语音，即将扬声器阵列的发声方向尽量调整到送话端定位出的用户说话的位置，即第二采集角度所确定的方向，以提高受话的私密性。具体是应用虚拟阵元技术，对扬声器SPK-b进行虚拟，虚拟扬声器SPK-c和扬声器SPK-a的连线指向语音方向，之后设置一个指向角度，将该第三采集角度的覆盖区域(即第三采集角度所包括的声音传播的区域范围)覆盖到第二采集角度确定的语音方向即可(如图5中所示SPK-a下方的角度张开范围覆盖了人头的位置)，优选的，该第三采集角度直接指向所述第二采集角度所述确定的语音方向。
参见图5，该免提通话设备还包括至少一个扬声器组成的受话端，在确定了 第二采集角度和方向完成送话端校准后，该方法还包括：对免提设备的受话端进行校准，设置受话端中一个扬声器的虚拟扬声器，该扬声器和其虚拟扬声器的连线指向γ1方向，并定义第三采集角度，第三采集角度指向第二采集角度确定的方向，第三采集角度实际上是播放声音的角度范围，该角度范围将第二采集角度和方向包括在内，使得扬声器尽可能准确的朝向语音源的方位播放，缩小播放语音范围，提高受话的私密性。
在免提通话设备的使用过程中，由于麦克风与扬声器之间的距离较近，因而可以使用送话端校准时确定的语音特征的方向来对受话端进行校准。
参见图5，设置辅扬声器SPK-b的虚拟扬声器SPK-c，在通过主扬声器SPK-a和γ1方向的反向延长线上设置虚拟扬声器SPK-c，虚拟扬声器SPK-c在以主扬声器SPK-a为圆心、主扬声器SPK-a和辅扬声器SPK-b连线为半径的弧形上，将主扬声器SPK-a和虚拟扬声器SPK-c组成新的阵列，定义新阵列的指向性角度为第三采集角度。将该第三采集角度指向送话端校准时确定的第二采集角度确定的方向完成受话端定位校准。
可以理解，附图3a、3b和图5中出现的角度是为了示意性的说明送话端和受话端定位时的角度范围，附图中出现的角度并非实际的角度大小。本发明实施例通过对送话端进行定位和校准，提高免提语音信号的信噪比，以及对受话端定位提高受话的私密性。
综上，本发明的这种提高免提设备通话质量的方法经过送话端校准和受话端校准后，既降低了周围环境噪声的干扰，提高送话信噪比，达到提高免提通话设备的通话质量的目的。又对扬声器阵列进行方向补偿，使其发出的声音刚好指向语音，提高了受话的私密性，增强用户使用体验。本方法可以应用在智能手表等具有免提通话功能的智能设备上，能大大提高智能设备通话性能。
根据本发明的另一个方面，提供了一种提高免提通话设备通话质量的装置；图6是本发明一个实施例提供的一种提高免提通话设备通话质量的装置的框图，该装置600包括：
语音特征确定单元601，用于在送话端初始的第一采集角度内进行扫描；当在第一采集角度内扫描到语音特征信号后，根据该语音特征信号的方向，在第一采集角度内确定一个比第一采集角度小的第二采集角度；
方向校准单元602，用于将送话端校准至该第二采集角度确定的方向。
其中，该装置还包括：虚拟麦克风创建单元，用于在送话端初始的第一采集角度内扫描到γ1方向有语音特征后，沿着γ1方向，经过主麦克做一条反向延长线，并以主麦克为圆心，以主麦克和其中一个辅麦克连线为半径画圆，圆弧与反向延长线相交的位置确定为主麦克的虚拟麦克风；
角度确认单元，用于以主麦克和主麦克的虚拟麦克风为新的语音阵列，并定义角度一个比第一采集角度小的β1，实时判断角度β1内是否有语音特征信号；是则，确认角度β1为第二采集角度；否则，以主麦克和其中一个辅麦克的连线为对称轴，定义角度β1关于对称轴的镜像角度β2，确认角度β2为第二采集角度。
其中，角度确认单元还用于，
对γ1方向的语音特征信号进行包络能量检测，当能量检测值大于第一预设阀值时，对γ1方向的语音特征信号进行过零率检测；
当γ1方向的语音特征信号的过零率达到第二预设阀值时，确认角度β1内包含语音特征信号。
其中，角度确认单元包括：
包络检测单元，用于通过如下公式对γ1方向的语音特征信号进行包络能量检测：
Figure PCTCN2015082634-appb-000010
其中，power为语音特征信号的能量值，参数alpha为计权因子，N为语音特征的具体的值；
过零检测单元，用于通过如下公式对γ1方向的语音特征信号进行过零率检测：
Figure PCTCN2015082634-appb-000011
其中，Z_rate为语音特征信号的过零率，n为离散时间序列中的一个值，
Figure PCTCN2015082634-appb-000012
其中，该装置还包括：受话端定位单元，用于设置受话端中一个扬声器的虚拟扬声器，该扬声器和其虚拟扬声器的连线指向γ1方向，并定义第三采集角度，所述第三采集角度所包括的区域范围包含所述第二采集角度确定的方向。
根据本发明的又一个方面，提供了一种免提通话设备，包括一个主麦克和至少一个辅麦克组成的送话端、至少一个扬声器组成的受话端以及上述的提高免提通话设备通话质量的装置。
需要说明的是，本发明的免提通话设备由于具有送话端和受话端，可以按照上述提高免提通话质量的方法在其受话端和送话端进行改进，但是在某些应用场合下，免提通话设备仅仅具有送话端或受话端，此时可以按照本发明具体实施例在受话端提高免提通话质量的方法来改进受话端或者在送话端提高免提通话质量的方法来改进送话端。即本发明的受话端和送话端的提高免提通话质量的方法可以单独分开实施，在此不再赘述。
以上所述仅为本发明的较佳实施例而已，并非用于限定本发明的保护范围。凡在本发明的精神和原则之内所作的任何修改、等同替换、改进等，均包含在本发明的保护范围内。

Claims (10)

  1. 一种提高免提通话设备通话质量的方法，所述免提通话设备包括一个主麦克和至少一个辅麦克组成的送话端，其特征在于，该方法包括：
    在送话端初始的第一采集角度内进行扫描；
    当在所述第一采集角度内扫描到语音特征信号后，根据该语音特征信号的方向，在所述第一采集角度内确定一个比所述第一采集角度小的第二采集角度；
    将所述送话端校准至该第二采集角度确定的方向。
  2. 如权利要求1所述的方法，其特征在于，所述在第一采集角度内扫描到的语音特征信号后，根据该语音特征信号的方向，在第一采集角度内确定一个比初始采集角度小的第二采集角度包括：
    在送话端初始的所述第一采集角度内扫描到γ1方向有语音特征信号后，沿着γ1方向，经过所述主麦克做一条反向延长线，并以所述主麦克为圆心，以所述主麦克和其中一个辅麦克连线为半径画圆，所述圆弧与所述反向延长线相交的位置确定为所述主麦克的虚拟麦克风；
    以所述主麦克和所述主麦克的虚拟麦克风为新的语音阵列，并定义一个比所述第一采集角度小的角度β1，实时判断所述角度β1内是否有语音特征信号；是则，确认所述角度β1为所述第二采集角度；
    否则，以所述主麦克和其中一个辅麦克的连线为对称轴，定义角度β1关于所述对称轴的镜像角度β2，确认所述角度β2为所述第二采集角度。
  3. 如权利要求书2所述的方法，其特征在于，所述实时判断所述角度β1内是否有语音特征信号包括：
    对γ1方向的语音特征信号进行包络能量检测，当所述能量检测值大于第一预设阀值时，对所述γ1方向的语音特征信号进行过零率检测；
    当所述γ1方向的语音特征信号的过零率达到第二预设阀值时，确认所述角度β1内有语音特征信号。
  4. 如权利要求书3所述的方法，其特征在于，所述对γ1方向的语音特征 进行包络能量检测包括：
    通过如下公式进行包络能量检测：
    Figure PCTCN2015082634-appb-100001
    其中，power为所述语音特征信号的能量值，所述参数alpha为计权因子，所述参数N为某一时点上的语音特征信号；
    所述对所述γ1方向的语音特征信号进行过零率检测包括：通过如下公式进行过零检测：
    Figure PCTCN2015082634-appb-100002
    其中，Z_rate为语音特征信号的过零率，n为离散时间序列中的一个值，
    Figure PCTCN2015082634-appb-100003
  5. 如权利要求书2所述的方法，所述免提通话设备还包括至少一个扬声器组成的受话端，其特征在于，该方法还包括：
    设置所述受话端中一个扬声器的虚拟扬声器，该扬声器和其虚拟扬声器的连线指向所述γ1方向，并定义第三采集角度，所述第三采集角度所包括的区域范围包含所述第二采集角度确定的方向。
  6. 一种提高免提通话设备通话质量的装置，其特征在于，该装置包括：
    语音特征确定单元，用于在送话端初始的第一采集角度内进行扫描；当在所述第一采集角度内扫描到语音特征信号后，根据该语音特征信号的方向，在所述第一采集角度内确定一个比所述第一采集角度小的第二采集角度；
    方向校准单元，用于将所述送话端校准至该第二采集角度确定的方向。
  7. 如权利要求6所述的装置，其特征在于，该装置还包括：
    虚拟麦克风创建单元，用于在送话端初始的所述第一采集角度内扫描到γ1方向有语音特征后，沿着γ1方向，经过所述主麦克做一条反向延长线，并以所述主麦克为圆心，以所述主麦克和其中一个辅麦克连线为半径画圆，所述圆弧与所述反向延长线相交的位置确定为所述主麦克的虚拟麦克风；
    角度确认单元，用于以所述主麦克和所述主麦克的虚拟麦克风为新的语音阵列，并定义一个比所述第一采集角度小的角度β1，实时判断所述角度β1内是否有语音特征；是则，确认所述角度β1为所述第二采集角度；否则，以所述主麦克和其中一个辅麦克的连线为对称轴，定义角度β1关于所述对称轴的镜像角度β2，确认所述角度β2为所述第二采集角度。
  8. 如权利要求7所述的装置，其特征在于，所述角度确认单元还用于：
    对γ1方向的语音特征进行包络能量检测，当所述能量检测值大于第一预设阀值时，对所述γ1方向的语音特征进行过零率检测；
    当所述γ1方向的语音特征的过零率达到第二预设阀值时，确认所述角度β1内包含语音特征；
    所述角度确认单元包括：
    包络检测单元，用于通过如下公式对γ1方向的语音特征信号进行包络能量检测：
    Figure PCTCN2015082634-appb-100004
    其中，power为语音特征信号的能量值，所述参数alpha为计权因子，所述N为语音特征的具体的值；
    过零检测单元，用于通过如下公式对所述γ1方向的语音特征信号进行过零率检测：
    Figure PCTCN2015082634-appb-100005
    其中，Z_rate为语音特征信号的过零率，n为离散时间序列中的一个值，
    Figure PCTCN2015082634-appb-100006
  9. 如权利要求书7所述的装置，其特征在于，该装置还包括：
    受话端定位单元，用于设置所述受话端中一个扬声器的虚拟扬声器，该某一扬声器和其虚拟扬声器的连线指向所述γ1方向，并定义第三采集角度，所述第三采集角度所包括的区域范围包含所述第二采集角度确定的方向。
  10. 一种免提通话设备，其特征在于，该免提通话设备包括一个主麦克和至少一个辅麦克组成的送话端、至少一个扬声器组成的受话端以及如权利要求6至9中任一项所述的装置。
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