WO2011153904A1 - Procédé de traitement de signaux vocaux et dispositif basé sur une matrice de microphones - Google Patents

Procédé de traitement de signaux vocaux et dispositif basé sur une matrice de microphones Download PDF

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Publication number
WO2011153904A1
WO2011153904A1 PCT/CN2011/074794 CN2011074794W WO2011153904A1 WO 2011153904 A1 WO2011153904 A1 WO 2011153904A1 CN 2011074794 W CN2011074794 W CN 2011074794W WO 2011153904 A1 WO2011153904 A1 WO 2011153904A1
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Prior art keywords
signal
microphone
sampling point
weight
speech signal
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PCT/CN2011/074794
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English (en)
Chinese (zh)
Inventor
何宏森
黄志宏
邱小军
袁浩
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中兴通讯股份有限公司
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Publication of WO2011153904A1 publication Critical patent/WO2011153904A1/fr

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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
    • 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 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 voice signal processing technologies, and in particular, to a voice signal processing method and apparatus based on a microphone array. Background technique
  • the multi-channel speech enhancement algorithm based on the microphone array combines the spatio-temporal information of the signal, and uses the difference of noise and speech to denoise.
  • it has become an important technology relying on multimedia conference, communication, voice control and other systems.
  • the sound quality and performance will seriously affect the overall effect and market competitiveness of the audio conferencing system. Therefore, for noise, noise cancellation is often achieved through the microphone array technology, which makes the participants of the audio conferencing system completely free from the handheld microphone and oriented.
  • the shackles of the microphone greatly improve the practicality of the audio conferencing system.
  • speech signal processing it is necessary to strive to make the speech quality of the input encoder better, such as low reverberation, low noise, etc., and the microphone array is designed to ensure low reverberation and low noise of the speech signal.
  • a "voice conference system” is disclosed in Chinese Patent Application Publication No. CN101496417A, the disclosure of which is hereby incorporated by reference.
  • a signal and thereafter, a signal level of the voice ⁇ bundle signal corresponding to the direction of arrival of the voice becomes higher, and the voice concentrating portion selects a voice ⁇ bundle signal whose signal level exceeds the set threshold, and sends the signal to the communication portion .
  • the main object of the present invention is to provide a method and apparatus for processing a voice signal based on a microphone array.
  • the strong directional microphone array can amplify the voice signal closest to the speaker, thereby dynamically tracking the speaker.
  • a voice signal processing method based on a microphone array the microphone array being composed of two or more directional microphones; the method comprising:
  • the determining, according to the energy value, an adjustment parameter of each voice signal of the same frame is:
  • the exponential adjustment processing is performed on each quotient value, and as an adjustment parameter of each voice signal, it is:
  • the E-th power of each quotient is used as an adjustment parameter of each speech signal; wherein E is a positive number greater than or equal to 2 and less than or equal to 10.
  • the determining, according to the adjustment parameter of each voice signal, the weight of each sampling point signal in the voice signal which is calculated according to the following formula:
  • w i ⁇ n w i ⁇ n- ⁇ ) + ⁇ -X)C-where, w ) is the weight of the nth sample point signal in the current speech signal frame in the microphone i, W -1) The weight of the n-1th sampling point signal in the current speech signal frame in the microphone i; is a predetermined forgetting factor, 0 ⁇ 1; C is an adjustment parameter of the current speech signal frame.
  • the determining, according to the adjustment parameter of each voice signal, the weight of each sampling point signal in the voice signal is:
  • w i ⁇ n w i ⁇ n- ⁇ ) + ⁇ -X)C-where, w ) is the initial weight of the nth sample point signal in the current speech signal frame in the microphone i, W -1) The initial weight of the n-1th sampling point signal in the current speech signal frame in the microphone i; is a predetermined forgetting factor, 0 ⁇ 1; C is an adjustment parameter of the current speech signal frame;
  • ⁇ ( ⁇ ) ⁇ ( (, ' , , --, where max ( ) is the maximum value.
  • the microphone array is a circular array or a spherical array; the number of microphones in the microphone array is 4 to 16.
  • a voice signal processing device based on a microphone array, the microphone array being composed of two or more directional microphones; the device comprising a first determining unit, a second determining unit, and a meter Computing unit and output unit;
  • a first determining unit configured to determine an energy value of a voice signal of the same frame received by each directional microphone
  • a second determining unit configured to determine, according to the energy value, an adjustment parameter of each voice signal of the same frame
  • a calculating unit configured to determine, according to an adjustment parameter of each voice signal, a weight of each sampling point signal in the voice signal, multiply each sampling point signal in each voice signal by a respective weight, and corresponding sampling points of each voice signal The product value of the signal is accumulated;
  • the output unit is configured to sequentially output the accumulated sampling point signals.
  • the second determining unit further compares the energy values of the voice signals of the same frame with the maximum energy value; and performs exponential adjustment processing on each quotient value as an adjustment parameter of each voice signal.
  • the second determining unit further uses the E-th power of each quotient as an adjustment parameter of each voice signal; wherein, E is a positive number greater than or equal to 2 and less than or equal to 10.
  • the calculating unit further calculates a weight of each sampling point signal in the voice signal according to the following formula:
  • w i ⁇ n w i ⁇ n- ⁇ ) + ⁇ -X)C-where, w ) is the weight of the nth sample point signal in the current speech signal frame in the microphone i, W -1) The weight of the n-1th sampling point signal in the current speech signal frame in the microphone i; is a predetermined forgetting factor, 0 ⁇ 1; C is an adjustment parameter of the current speech signal frame.
  • the calculating unit further calculates the weight of each sampling point signal in the voice signal as follows:
  • w i ⁇ n w i ⁇ n- ⁇ ) + ⁇ -X)C-where, w ) is the initial weight of the nth sample point signal in the current speech signal frame in the microphone i, W -1) Is the initial weight of the n-1th sampling point signal in the current speech signal frame in the microphone i; is a predetermined forgetting factor, 0 ⁇ 1; C The adjustment parameter for the current speech signal frame;
  • ⁇ (n) ⁇ ( (, 1 —- , where max ( ) is the maximum value.
  • the microphone array is a circular array or a spherical array; the number of microphones in the microphone array is 3 to 16.
  • a strong array of N directional microphones is used to form a circular array, and the pickup of the array covers a 360-degree orientation; first, the energy value of the speech signal received by each microphone in the strong directional microphone array, and the energy of the speech signal.
  • the value information determines an adjustment parameter of the voice signal of the current voice frame received by each microphone, and uses the adjustment parameter to calculate the weight of each sample point signal of the current voice frame, and the calculated weight value and the corresponding weight
  • the sample signals are multiplied, and the product of the sample signals at the same position is accumulated and sequentially output in the order of the sample points.
  • the invention utilizes the energy values of the speech signals received by the microphones in the microphone array to determine the adjustment parameters of the respective speech signals, and smoothes the signal of each sample point by using the forgetting factor, so that the outputted speech signals are more consistent.
  • the invention has simple calculation method, does not require complicated calculations and circuits, and has good anti-reverberation and directional pickup functions.
  • FIG. 1 is a flow chart of a method for processing a voice signal based on a microphone array according to the present invention
  • FIG. 2 is a normalized energy change of a voice signal frame of a voice signal picked up by each microphone in a microphone array when two sound sources of the reverberation chamber are switched to each other. Schematic diagram of the relationship;
  • FIG. 3 is a schematic diagram showing the relationship between the average weights of the voice frames of each channel in the output signal of the microphone array when the two sound sources are switched to each other in the reverberation chamber;
  • FIG. 4 is a schematic diagram showing a normalized energy change relationship of a speech signal speech frame picked up by each microphone in the microphone array when two sound sources are simultaneously sounded in the reverberation chamber
  • FIG. 5 is a schematic diagram showing the relationship between the average weights of voice frames of each channel in the output signal of the microphone array when two sound sources are simultaneously sounded in the reverberation chamber;
  • FIG. 6 is a schematic diagram showing a normalized energy change relationship of a speech signal speech frame picked up by each microphone in the microphone array when two sound sources are switched to each other in an ordinary room;
  • FIG. 7 is a schematic diagram showing the relationship between the average weights of the voice frames of each channel in the output signal of the microphone array when the two sound sources are switched to each other in an ordinary room;
  • FIG. 8 is a schematic diagram showing a normalized energy change relationship of a speech signal speech frame picked up by each microphone in the microphone array when two sound sources are simultaneously sounded in an ordinary room;
  • FIG. 9 is a schematic diagram showing the relationship between the average weights of voice frames of each channel in the output signal of the microphone array when two sound sources are simultaneously sounded in an ordinary room;
  • Fig. 10 is a schematic view showing the structure of a voice signal processing apparatus based on a microphone array of the present invention. detailed description
  • the basic idea of the present invention is to form a circular array by using N strong directional microphones, so that the pickup of the microphone array covers 360 degrees of orientation; calculate the energy of the signals picked up by the microphones, and maintain the maximum energy by comparing the energy.
  • the amplitude of the speech signal of the channel is unchanged, and the speech signal of other channels is weakened; the degree of weakening of the speech signal is controlled by the adjustment parameter; and, in order to ensure that the speech signal is smooth and natural without switching noise when switching between channels based on energy comparison, A smoothing mechanism-forgetting factor is introduced, and the current sampling point is combined with the signal of the previous sampling point to switch.
  • the microphones in the microphone array are all strong directivity microphones, rather than omnidirectional microphones.
  • the so-called strong directional microphone that is, the microphone can perform the collection of voice signals by pointing. Strong directional microphone can effectively reduce the reverberation intensity entering each microphone; the invention It is the directional pick-up feature of the strong directional microphone that uses the energy of the same speech frame picked up by each microphone to determine the weight of each sample signal in each of the same speech frames, so that the output is better. Voice signal.
  • the microphone array of the present invention uses a circumferential or spherical layout to collect the speech signals of the various bits.
  • the number of strong directional microphones in the microphone array is generally 3 to 16, so as to be evenly distributed on the set circumference or the spherical surface, and the corresponding microphones are provided for the respective points to perform voice collection.
  • the radius of the circumference or the sphere is generally 3 to 20 cm, and the diaphragms of the microphones face outward in the radial direction of the circumference or the sphere.
  • FIG. 1 is a flowchart of a method for processing a voice signal based on a microphone array according to the present invention.
  • the energy value of the first frame signal received by the (1, 2, . . . , N) microphones is as shown in equation (2):
  • the length of the speech frame of each channel for calculating energy can be taken as 400 ms; the system response time of adaptive switching between channels is taken as 400 ms.
  • the above frame length is processed by The processing speed of the device is determined, and other lengths, such as 450ms or 500ms, can also be taken.
  • Step 102 Normalize the energy value determined by the equation (2) based on the maximum value of the energy of the frame signal of the N channels.
  • the normalization process is to convert the energy value of the frame signal of each channel to a value between 0 and 1, for subsequent processing. Normalization
  • the purpose of determining the adjustment parameters is to make the speech signal on the channel with a large energy value larger, and to make the speech signal on the channel with a smaller energy value smaller, and thereby increase the energy value and the larger speech signal and energy.
  • the difference between the smaller voice signals, which can highlight the signal in the direction of the sound source, suppress the signal in other directions, make the sound clearer and the reverberation is smaller.
  • the selected adjustment index value is a positive number greater than or equal to 2 and less than or equal to 10.
  • the adjustment index is generally selected 4, 5, 6.
  • the adjustment parameter WW is determined as shown in equation (4):
  • the adjustment index adjusts the proportion of each channel signal in the output signal according to the energy relationship of the speech frames of each channel.
  • Step 104 Calculate weights of the nth sample point signals of the (1, 2, .., N) microphone sets in the array output signal; the change of the weight is according to each sample point signal
  • the forgetting factor is used to smooth the volume of the speech frame before and after switching, to avoid the flickering of the speech signal, and to suppress the switching noise caused by the change of the speech frame energy of the channel when switching.
  • the specific value is determined by the smoothness desired by the user.
  • Step 106 Calculate the output sample signal of the microphone array, and output them in sequence.
  • the output of each sample signal is as shown in equation (7):
  • the corresponding sample point signals of the respective microphones are accumulated as the output sample point signals.
  • the typical front-end processing before entering the processing of the algorithm in actual work is to convert the voice signal into an electrical signal through a microphone, and perform processing by amplifying and analog-to-digital conversion into a digital signal processor (DSP).
  • DSP digital signal processor
  • the microphone array is evenly distributed along the circumference of four microphones to illustrate the results of speech signal processing in each application environment.
  • the radius of the circumference is 5cm
  • FIG. 2 is a schematic diagram showing the relationship between the normalized energy changes of the speech signal speech frames picked up by the microphones in the microphone array when the two sound sources are switched to each other in the reverberation chamber, as shown in FIG. 2, showing the reverberation chamber.
  • the normalized energy change relationship of the speech signal speech frames picked up by the microphones in the microphone array is calculated by the method of the present invention after calculating the energy of the speech frames picked up by the respective microphones.
  • FIG. 3 is a schematic diagram showing the relationship between the average weights of the voice frames of each channel in the output signal of the microphone array when the two sound sources are switched to each other in the reverberation chamber, as shown in FIG. 3, two sound sources in the reverberation chamber.
  • the average weight value of the voice frames of each channel in the output signal of the microphone array is calculated by the method of the present invention, and the present invention can be picked up according to the microphones.
  • the voice frame energy of the sound is automatically switched, and the switching process is naturally stable.
  • the voice signal picked up by each microphone is processed by the method of the present invention, the sound quality of the output voice signal of the microphone array is smooth and natural, and the reverberation is greatly reduced.
  • FIG. 4 is a schematic diagram showing the relationship between the normalized energy changes of the speech signal speech frames picked up by the microphones in the microphone array when the two sound sources are simultaneously sounded in the reverberation chamber, as shown in FIG. 4, which shows two in the reverberation chamber.
  • the method of the present invention calculates the normalized energy change relationship of the speech frame energy picked up by each microphone and the speech signal speech frame picked up by each microphone in the microphone array.
  • FIG. 5 is a schematic diagram showing the relationship between the average weights of the speech frames of each channel in the output signal of the microphone array when two sound sources are simultaneously sounded in the reverberation chamber, as shown in FIG. 5, at the same time in the reverberation chamber.
  • the method of the present invention calculates the average frame weight change of the speech frame energy of each channel in the output signal of the microphone array. It can be seen that the present invention can automatically switch according to the size of the speech frame energy of each microphone pickup, and the switching process is naturally stable. After the voice signal picked up by each microphone is processed by the method of the present invention, the sound quality of the output voice signal of the microphone array is smooth and natural.
  • FIG. 6 is a schematic diagram showing the relationship of the normalized energy change of the speech signal speech frames picked up by the microphones in the microphone array when the two sound sources are switched to each other in an ordinary room. As shown in FIG. 6, two are shown in the ordinary room.
  • the method of the present invention calculates the normalized energy change relationship of the speech frame energy picked up by each microphone and the speech signal speech frame picked up by each microphone in the microphone array.
  • FIG. 7 is a schematic diagram showing the relationship between the average weights of voice frames of each channel in the output signal of the microphone array when two sound sources are switched to each other in an ordinary room, as shown in FIG.
  • the method of the present invention is used to calculate the energy of the speech frame picked up by each microphone, and the average weight change relationship of the voice frames of each channel in the output signal of the microphone array. It can be seen that the present invention can automatically switch according to the size of the speech frame energy of each microphone pickup, and the switching process is naturally stable. After the voice signal picked up by each microphone is processed by the method of the present invention, the sound quality of the output voice signal of the microphone array is smooth and natural. Reverberation is reduced.
  • FIG. 8 is a schematic diagram showing the relationship between the normalized energy changes of the speech signal speech frames picked up by the microphones in the microphone array when two sound sources are simultaneously sounded in an ordinary room. As shown in FIG. 8, two sound sources simultaneously sound in the ordinary room.
  • the present invention is used to calculate the speech frame energy picked up by each microphone, and the normalized energy change relationship of the speech signal speech frames picked up by the microphones in the microphone array;
  • FIG. 9 is a schematic diagram showing the relationship between the average weights of voice frames of each channel in the output signal of the microphone array when two sound sources are simultaneously sounded in an ordinary room. As shown in FIG. 9, when two sound sources are simultaneously sounded in an ordinary room.
  • the invention calculates the average frame weight change of the speech frame energy of each channel in the output signal of the microphone array by using the present invention. It can be seen that the present invention can automatically switch according to the size of the speech frame energy of each microphone pickup, and the switching process is naturally stable. After the voice signal picked up by each microphone is processed by the method of the present invention, the sound quality of the output voice signal of the microphone array is smooth and natural.
  • the speech signal processed by the above steps can be output as a digital signal or as an analog signal after digital-to-analog conversion.
  • FIG. 10 is a schematic structural diagram of a structure of a voice signal processing apparatus based on a microphone array according to the present invention. As shown in FIG. 10, the apparatus includes a first determining unit 100, a second determining unit 101, a calculating unit 102, and an output unit 103.
  • a first determining unit 100 configured to determine an energy value of a voice signal of the same frame received by each directional microphone
  • a second determining unit 101 configured to determine, according to the energy value, an adjustment parameter of each voice signal of the same frame
  • the calculating unit 102 is configured to determine weights of the sampling point signals in the voice signal according to the adjustment parameters of the voice signals, multiply each sampling point signal in each voice signal by a respective weight, and perform corresponding sampling on each voice signal.
  • the product value of the point signal is accumulated;
  • the output unit 103 is configured to sequentially output the accumulated sampling point signals.
  • the microphone array is composed of two or more directional microphones.
  • the second determining unit 101 further compares the energy values of the speech signals of the same frame with the maximum energy value; and performs exponential adjustment processing on each quotient value as an adjustment parameter of each speech signal.
  • the second determining unit 101 further uses the E-th power of each quotient value as an adjustment parameter of each speech signal; wherein E is a positive number greater than or equal to 2 and less than or equal to 10.
  • W -1) is the weight of the n-1th sampling point signal in the current speech signal frame in the microphone i; is a predetermined forgetting factor, 0 ⁇ 1; C is the adjustment parameter of the current speech signal frame.
  • the above calculating unit 102 further calculates the weight of each sampling point signal in the speech signal as follows:
  • w,(n) /lw,(nl) + (l-/l)C;
  • w ) is the initial weight of the nth sample point signal in the current speech signal frame in the microphone i, W -1)
  • the initial weight of the n-1th sampling point signal in the current speech signal frame in the microphone i; is a predetermined forgetting factor, 0 ⁇ 1;
  • C is an adjustment parameter of the current speech signal frame;
  • ⁇ (n) ⁇ ( (, ⁇ , ---, where max ( ) is the maximum value.
  • the microphone array described above is a circular array or a spherical array; the number of microphones in the microphone array is 3 to 16.
  • the microphone signal processing apparatus based on the microphone array shown in FIG. 10 is designed to implement the aforementioned voice signal processing method based on the microphone array, and the functions of the processing units in the apparatus shown in FIG. 10 can be referred to. It is understood from the description of the foregoing method that the functions of the various processing units can be implemented by a program running on a processor, or by a specific logic circuit.

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

Abstract

La présente invention concerne un procédé de traitement de signaux vocaux basé sur une matrice de microphones, et la matrice de microphones est composée de plus de deux microphones directionnels. Le procédé comprend les étapes suivantes consistant : à déterminer les valeurs d'énergie des signaux vocaux de la même trame, reçues par chaque microphone directionnel; à déterminer les paramètres d'ajustement de signaux vocaux de la même trame en fonction des valeurs d'énergie; à déterminer le poids de chaque signal de point d'échantillonnage dans les signaux vocaux en fonction du paramètre d'ajustement de chaque signal vocal, à multiplier chaque signal de point d'échantillonnage dans chaque signal vocal par chaque poids, à accumuler les valeurs de produit des signaux de point d'échantillonnage correspondant à chaque signal vocal et à produire séquentiellement les signaux de point d'échantillonnage accumulés. La présente invention concerne aussi un dispositif de traitement de signaux vocaux basé sur la matrice de microphones. La présente invention a un mode de calcul simple, ne nécessite aucun calcul ni circuit complexe, et a une résistance à la réverbération et des fonctions d'acquisition orientée avantageuses.
PCT/CN2011/074794 2010-06-08 2011-05-27 Procédé de traitement de signaux vocaux et dispositif basé sur une matrice de microphones WO2011153904A1 (fr)

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CN101867853B (zh) * 2010-06-08 2014-11-05 中兴通讯股份有限公司 基于传声器阵列的语音信号处理方法及装置
CN103124386A (zh) * 2012-12-26 2013-05-29 山东共达电声股份有限公司 一种远讲用降噪、消回波、锐指向传声器
US9674607B2 (en) * 2014-01-28 2017-06-06 Mitsubishi Electric Corporation Sound collecting apparatus, correction method of input signal of sound collecting apparatus, and mobile equipment information system
CN105652243B (zh) * 2016-03-14 2017-12-05 西南科技大学 多通道群稀疏线性预测时延估计方法
CN110570874B (zh) * 2018-06-05 2021-10-22 中国科学院声学研究所 一种用于监测野外鸟类鸣声强度及分布的系统及其方法

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WO2009042948A1 (fr) * 2007-09-28 2009-04-02 Qualcomm Incorporated Détecteur d'activité vocale à microphones multiples
CN101658052A (zh) * 2007-03-21 2010-02-24 弗劳恩霍夫应用研究促进协会 用于音频重构增强的方法和设备
CN101867853A (zh) * 2010-06-08 2010-10-20 中兴通讯股份有限公司 基于传声器阵列的语音信号处理方法及装置

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WO2009009568A2 (fr) * 2007-07-09 2009-01-15 Mh Acoustics, Llc Ensemble de microphones elliptiques augmentés
WO2009042948A1 (fr) * 2007-09-28 2009-04-02 Qualcomm Incorporated Détecteur d'activité vocale à microphones multiples
CN101867853A (zh) * 2010-06-08 2010-10-20 中兴通讯股份有限公司 基于传声器阵列的语音信号处理方法及装置

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