CN109959893A - A kind of acoustical signal angle estimating method based on Beidou time service and microphone array - Google Patents
A kind of acoustical signal angle estimating method based on Beidou time service and microphone array Download PDFInfo
- Publication number
- CN109959893A CN109959893A CN201910146630.2A CN201910146630A CN109959893A CN 109959893 A CN109959893 A CN 109959893A CN 201910146630 A CN201910146630 A CN 201910146630A CN 109959893 A CN109959893 A CN 109959893A
- Authority
- CN
- China
- Prior art keywords
- array element
- sound source
- signal
- angle
- array
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Pending
Links
Classifications
-
- G—PHYSICS
- G01—MEASURING; TESTING
- G01S—RADIO DIRECTION-FINDING; RADIO NAVIGATION; DETERMINING DISTANCE OR VELOCITY BY USE OF RADIO WAVES; LOCATING OR PRESENCE-DETECTING BY USE OF THE REFLECTION OR RERADIATION OF RADIO WAVES; ANALOGOUS ARRANGEMENTS USING OTHER WAVES
- G01S3/00—Direction-finders for determining the direction from which infrasonic, sonic, ultrasonic, or electromagnetic waves, or particle emission, not having a directional significance, are being received
- G01S3/80—Direction-finders for determining the direction from which infrasonic, sonic, ultrasonic, or electromagnetic waves, or particle emission, not having a directional significance, are being received using ultrasonic, sonic or infrasonic waves
- G01S3/802—Systems for determining direction or deviation from predetermined direction
Landscapes
- Physics & Mathematics (AREA)
- Engineering & Computer Science (AREA)
- General Physics & Mathematics (AREA)
- Radar, Positioning & Navigation (AREA)
- Remote Sensing (AREA)
- Circuit For Audible Band Transducer (AREA)
Abstract
The invention discloses a kind of acoustical signal angle estimating method based on Beidou time service and microphone array, it can be to environment noise-less pollution, it effectively solves the problems, such as the noise and reverberation that acoustical signal occurs in communication process, obtains angle information of the sound source with respect to base station.The hardware device that the invention is embedded in beidou timing module by two is constituted, and sound source is the loudspeaker of capable of emitting customization short duration high frequency linear frequency modulation acoustical signal (chirp signal), and base station is the microphone array that multiple microphones are constituted.Firstly, sound source is synchronous with base station progress Microsecond grade big dipper clock.Then, sound source sends chirp signal, is received by base station and used GCC-PHAT algorithm and reverberation filtering algorithm to obtain the distance between sound source and base station information.Finally array element institute ranging deviation is converted with geometric method and is at an angle of, and final angle of the sound source with respect to base station is obtained using weighted sum mode.
Description
Technical field
The invention belongs to Beidous and acoustics alignment by union field, more particularly to one kind to be based on Beidou time service and microphone array
Acoustical signal angle estimating method.
Background technique
Currently, high accuracy positioning demand increasingly increases, GNSS positioning causes positioning to be failed vulnerable to blocking for building,
Therefore it needs using the positioning in additional reference signal ancillary chamber with complicated outdoor environment.In the prior art, UWB, WIFI and
The methods of Bluetooth due to the factor constraints such as its own compatibility, ranging range, range accuracy big mould popularization in more
It is difficult.It is transported extensively in fields such as radar, sonars as a kind of high-precision positioning method based on the location technology of acoustics
With strong compatibility, low cost, the important means for being increasingly becoming positioning field in high precision.
For the angle estimating method of high frequency sound signal, traditional MUSIC algorithm is unable to satisfy half-wave due to array element interval
Length requires and angle of arrival ambiguity solution problem, and the algorithm based on time delay is easy to be influenced and generate larger by ambient noise
Deviation, and effective solution there is no for multipath reverberation situation.The high-frequency chirp acoustical signal of 18KHz-22KHz belongs to people
The signal frequency range that ear is not heard is smaller to the noise and interference of environment generation in communication process.The signal propagation distance is remote, by
Environment influence is small, is suitable as the basis signal of acoustics positioning.It, can by big dipper clock synchronization and microphone array signal processing
The influence for effectively weakening the environment number of making an uproar, reverberation, improves ranging and angle measurement accuracy, shows more excellent performance.
Summary of the invention
In view of the deficienciess of the prior art, Beidou time service and microphone array can be utilized it is an object of the invention to provide a kind of
Equipment, the method for doing angle estimation using high frequency sound signal under noise and reverberant ambiance.
To achieve the above object, the present invention provides a kind of acoustical signal ranging based on Beidou time service and microphone array and is used in combination
In angle estimating method, this method is made of the loudspeaker of insertion beidou timing module and microphone speaker array, feature
It is, comprising the following steps:
Step 1: clock is synchronous: loudspeaker and microphone speaker array utilize embedded beidou timing module realization microsecond
Grade clock is synchronous；
Step 2: ranging: loudspeaker is periodically sent out short duration high frequency linear frequency modulation acoustical signal, and microphone speaker array is each
Array element acquires acoustical signal, is sent and received signal by GCC-PHAT algorithm and the processing of reverberation filtering algorithm, obtains sound source and battle array
The distance between member information, specific implementation step are as follows:
(1a) is reference with any period, and loudspeaker is from t0Moment issues short duration high frequency linear frequency modulation acoustical signal s
(t) (also referred to as chirp signal), frequency range 18KHz-22KHz belong to the non-audible frequency range of human ear；
(1b) microphone speaker array shares M array element, each array element m1,m2,…,mMRespectively from t1,t2,…,tMMoment
It receives through overdamping, the chirp acoustical signal of reflection and ambient noise signal:
ym(t)=αm(t)×s(t-Δtm)*hm(t)+nm(t), m=1,2 ..., M (1)
Wherein, symbol * indicates linear convolution, Δ tm=tm-t0, m=1,2 ..., M indicates that chirp signal is direct from sound source
Travel to the practical duration of m-th of array element, ym(t) indicate m-th of array element in the received resultant signal of t moment, s (t- Δ tm) indicate
Sound source is in t- Δ tmThe signal that moment issues, nm(t) indicate m-th of array element in the received ambient noise signal of t moment, αm(t) table
Show that m-th of array element receives the decay factor of sound-source signal, h in t momentm(t) indicate that m-th of array element receives sound source letter in t moment
Number channel impulse response；
(1c) is by ym(t) and s (t) carries out Fast Fourier Transform (FFT), and time domain t is converted into frequency domain ω, respectively obtains Ym(ω)
With S (ω).Then GCC-PHAT algorithm is used to transformed signal, solves the function about duration τ:
Wherein, S*(ω) indicates the adjoint matrix of S (ω),Indicate YmThe relevance function of (ω) and S (ω), τm
Indicate that sound source issues chirp signal and reaches m-th of array element duration；
(1d) in view of acoustical signal indoors in communication process by wall and barrier reflection and there is multipath effect,
Therefore practical duration is less than or equal to duration required by GCC-PHAT, it may be assumed that
Δtm=tm-t0≤τm, m=1,2 ..., M (4)
In order to reduce the influence of multipath reverberation, using reverberation filtering algorithm: for m-th of array element, with when a length of axis, from a left side
To the right, first wave crest that relevance function meets the following conditions, the duration at the peak are foundIt is approximately equal to Δ tm:
Wherein, τ0For time interval constant, λ ∈ (0,1) indicates constant related with environment reverberation degree；
Temporal information is converted into the range information of array element and sound source by (1e):
Wherein, xmIt is m-th of array element at a distance from sound source, c is the velocity of sound.
Step 3: the range information of array element and sound source is converted into range difference information using geometric method, in conjunction with array element by angle measurement
Spacing distance calculate angle information, summation is weighted to each angle, obtains the angle of sound source facing arrays, specific steps
It is as follows:
(2a) is directed to far-field signal, is estimated using geometric method sound source angle, by each array element and sound source range information
It is converted into range difference information:
xmn=xm-xn (7)
Wherein, xmnIndicate the range difference of m-th of array element Yu n-th array element and sound source distance；
Range difference information is converted into angle information by (2b):
Wherein, θmnIndicate m-th of array element and the comprehensive angle obtained of n-th of array element, dmnIndicate m-th of array element and n-th
Spacing distance of the array element on array；
(2c) is angled to institute to be weighted, and the final angle of sound source opposing microphones array is obtained
Wherein, ηmnIndicate θmnShared weight is fixed constant, and meets ηmn∈ (0,1) and
Compared with prior art, the invention has the following advantages that
1. sound source and base station are mainly made of loudspeaker, microphone speaker array and beidou timing module, system is by north
The time service module that struggles against realizes that microsecond rank is synchronous, and synchronization time, precision met the demand of the ranging of acoustical signal, and equipment cost is low.
2. the linear frequency modulation acoustical signal for the 18KHz-22KHz that loudspeaker issues is more than the receivable frequency range of human ear, right
Ambient noise interference is small, and propagation distance is remote.And the frequency range compatible common microphone array and mobile phone terminal, it has a extensive future.
3. in conventional angle estimation method, carrying out angle estimation merely with array signal, easily going out under multipath reverberant ambiance
The problem of existing angular distortions.The GCC-PHAT algorithm and reverberation elimination algorithm that the present invention is used in ranging can effectively solve noise
With reverberation problem, range accuracy is promoted.In angle measurement based on range information with high precision, the geometry angle-measuring method and angle of use
The precision and robustness of angle estimation can be improved in weighting method.
Detailed description of the invention
Fig. 1 is system construction drawing；
Fig. 2 is array element and sound source ranging schematic diagram；
Fig. 3 is far-field signal angle schematic diagram；
Fig. 4 is geometric method measuring angle schematic diagram.
Specific embodiment
The following describes the present invention in detail with reference to the accompanying drawings and specific embodiments.
Referring to Fig.1, the present invention provides a kind of acoustical signal ranging based on Beidou time service and microphone array and for angle
Estimation method, this method are made of the loudspeaker of insertion beidou timing module and microphone speaker array, comprising the following steps:
Step 1: clock is synchronous: loudspeaker and microphone speaker array utilize embedded beidou timing module realization microsecond
Grade clock is synchronous；
Step 2: ranging: loudspeaker is periodically sent out short duration high frequency linear frequency modulation acoustical signal, and microphone speaker array is each
Array element acquires acoustical signal, is sent and received signal by GCC-PHAT algorithm and the processing of reverberation filtering algorithm, obtains sound source and battle array
The distance between member information, specific implementation step are as follows:
(1a) is reference with any period, and loudspeaker is from t0Moment issues short duration high frequency linear frequency modulation acoustical signal s
(t) (also referred to as chirp signal), frequency range 18KHz-22KHz belong to the non-audible frequency range of human ear；
(1b) microphone speaker array shares M array element, each array element m1,m2,…,mMRespectively from t1,t2,…,tMMoment
It receives through overdamping, the chirp acoustical signal of reflection and ambient noise signal:
ym(t)=αm(t)×s(t-Δtm)*hm(t)+nm(t), m=1,2 ..., M (1)
Wherein, symbol * indicates linear convolution, Δ tm=tm-t0, m=1,2 ..., M indicate that chirp signal is direct from sound source
Travel to the practical duration of m-th of array element, ym(t) indicate m-th of array element in the received resultant signal of t moment, s (t- Δ tm) indicate
Sound source is in t- Δ tmThe signal that moment issues, nm(t) indicate m-th of array element in the received ambient noise signal of t moment, αm(t) table
Show that m-th of array element receives the decay factor of sound-source signal, h in t momentm(t) indicate that m-th of array element receives sound source letter in t moment
Number channel impulse response；
(1c) is by ym(t) and s (t) carries out Fast Fourier Transform (FFT), and time domain t is converted into frequency domain ω, respectively obtains Ym(ω)
With S (ω).Then GCC-PHAT algorithm is used to transformed signal, solves the function about duration τ:
Wherein, S*(ω) indicates the adjoint matrix of S (ω),Indicate YmThe relevance function of (ω) and S (ω), τm
Indicate that sound source issues chirp signal and reaches m-th of array element duration；
(1d) in view of acoustical signal indoors in communication process by wall and barrier reflection and there is multipath effect,
Therefore practical duration is less than or equal to duration required by GCC-PHAT, it may be assumed that
Δtm=tm-t0≤τm, m=1,2 ..., M (4)
In order to reduce the influence of multipath reverberation, using reverberation filtering algorithm: for m-th of array element, with when a length of axis, from a left side
To the right, first wave crest that relevance function meets the following conditions, the duration at the peak are foundIt is approximately equal to Δ tm:
Wherein, τ0For time interval constant, λ ∈ (0,1) indicates constant related with environment reverberation degree；
Temporal information as shown in Fig. 2, is converted into the range information of array element and sound source by (1e):
Wherein, xmIt is m-th of array element at a distance from sound source, c is the velocity of sound.
Step 3: angle measurement, as shown in figure 3, the angle that sound incident direction and array vertical direction are formed is considered as incidence angle,
M-th of array element incidence angle is θm, in the range ofThe range information of array element and sound source is converted into using geometric method
Range difference information, calculates angle information in conjunction with the spacing distance of array element, is weighted summation to each angle, it is opposite to obtain sound source
The angle of array, the specific steps are as follows:
(2a) is directed to far-field signal, is estimated using geometric method sound source angle, by each array element and sound source range information
It is converted into range difference information:
xmn=xm-xn (7)
Wherein, xmnIndicate the range difference of m-th of array element Yu n-th array element and sound source distance；
Range difference information is converted into angle information by (2b):
Wherein, θmnIndicate m-th of array element and the comprehensive angle obtained of n-th of array element, dmnIndicate m-th of array element and n-th
Spacing distance of the array element on array.
As shown in figure 4, choosing the linear microphone array of M=5,10 groups of range difference information and array spacings distance can get
Information, to be convertible into 10 groups of angle informations；
(2c) is angled to institute to be weighted, and the final angle of sound source opposing microphones array is obtained
Wherein, ηmnIndicate θmnShared weight is fixed constant, and meets ηmn∈ (0,1) and
Claims (3)
1. a kind of acoustical signal angle estimating method based on Beidou time service and microphone array, which is characterized in that this method includes
Following steps:
Step 1: clock is synchronous: when loudspeaker and microphone speaker array realize Microsecond grade using embedded beidou timing module
Clock is synchronous.
Step 2: ranging: loudspeaker is periodically sent out short duration high frequency linear frequency modulation acoustical signal, each array element of microphone speaker array
Acquire acoustical signal, sent and received signal by GCC-PHAT algorithm and the processing of reverberation filtering algorithm, obtain sound source and array element it
Between range information.
Step 3: angle measurement: the range information of array element and sound source being converted by range difference information using geometric method, in conjunction between array element
Gauge is weighted summation from angle information is calculated, to each angle, obtains the angle of sound source facing arrays.
2. a kind of acoustical signal angle estimating method based on Beidou time service and microphone array according to claim 1,
It is characterized in that, the step 2 specifically includes following sub-step:
(1a) is reference with any period, and loudspeaker is from t0Moment, sending short duration high frequency linear frequency modulation acoustical signal s (t) (
Claim chirp signal), frequency range 18KHz-22KHz belongs to the non-audible frequency range of human ear；
(1b) microphone speaker array shares M array element, each array element m1,m2,…,mMRespectively from t1,t2,…,tMReception
To chirp acoustical signal and ambient noise signal through overdamping, reflection:
ym(t)=αm(t)×s(t-Δtm)*hm(t)+nm(t), m=1,2 ..., M (1)
Wherein, symbol * indicates linear convolution, Δ tm=tm-t0, m=1,2 ..., M indicate that chirp signal is directly propagated from sound source
To the practical duration of m-th of array element, ym(t) indicate m-th of array element in the received resultant signal of t moment, s (t- Δ tm) indicate sound source
In t- Δ tmThe signal that moment issues, nm(t) indicate m-th of array element in the received ambient noise signal of t moment, αm(t) the is indicated
M array element receives the decay factor of sound-source signal, h in t momentm(t) indicate that m-th of array element receives sound-source signal in t moment
Channel impulse response；
(1c) is by ym(t) and s (t) carries out Fast Fourier Transform (FFT), and time domain t is converted into frequency domain ω, respectively obtains Ym(ω) and S
(ω)；Then GCC-PHAT algorithm is used to transformed signal, solves the function about duration τ:
Wherein, S*(ω) indicates the adjoint matrix of S (ω), RYm,S(τ) indicates YmThe relevance function of (ω) and S (ω), τmIt indicates
Sound source issues chirp signal and reaches m-th of array element duration；
(1d) in view of acoustical signal indoors in communication process by wall and barrier reflection and there is multipath effect, therefore
Practical duration is less than or equal to duration required by GCC-PHAT, it may be assumed that
Δtm=tm-t0≤τm, m=1,2 ..., M (4)
In order to reduce the influence of multipath reverberation, using reverberation filtering algorithm；For m-th of array element, with when a length of axis, from left-hand
First wave crest that relevance function meets the following conditions, the duration at the peak are found in the right sideIt is approximately equal to Δ tm:
Wherein, τ0For time interval constant, λ ∈ (0,1) indicates constant related with environment reverberation degree；
Temporal information is converted into the range information of array element and sound source by (1e):
Wherein, xmIt is m-th of array element at a distance from sound source, c is the velocity of sound.
3. a kind of acoustical signal angle estimating method based on Beidou time service and microphone array according to claim 1,
It is characterized in that, the step 3 specifically includes following sub-step:
(2a) is directed to far-field signal, is estimated using geometric method sound source angle, and each array element and sound source range information are converted
At range difference information:
xmn=xm-xn(7)
Wherein, xmnIndicate the range difference of m-th of array element Yu n-th array element and sound source distance；
Range difference information is converted into angle information by (2b):
Wherein, θmnIndicate m-th of array element and the comprehensive angle obtained of n-th of array element, dmnIndicate m-th of array element and n-th of array element
Spacing distance on array；
(2c) is angled to institute to be weighted, and the final angle of sound source opposing microphones array is obtained
Wherein, ηmnIndicate θmnShared weight is fixed constant, and meets ηmn∈ (0,1) and
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201910146630.2A CN109959893A (en) | 2019-02-27 | 2019-02-27 | A kind of acoustical signal angle estimating method based on Beidou time service and microphone array |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201910146630.2A CN109959893A (en) | 2019-02-27 | 2019-02-27 | A kind of acoustical signal angle estimating method based on Beidou time service and microphone array |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CN109959893A true CN109959893A (en) | 2019-07-02 |
Family
ID=67023609
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201910146630.2A Pending CN109959893A (en) | 2019-02-27 | 2019-02-27 | A kind of acoustical signal angle estimating method based on Beidou time service and microphone array |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN109959893A (en) |
Cited By (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN110376551A (en) * | 2019-07-04 | 2019-10-25 | 浙江大学 | A kind of TDOA localization method based on the distribution of acoustical signal time-frequency combination |
Citations (8)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102707262A (en) * | 2012-06-20 | 2012-10-03 | 太仓博天网络科技有限公司 | Sound localization system based on microphone array |
| US20140241549A1 (en) * | 2013-02-22 | 2014-08-28 | Texas Instruments Incorporated | Robust Estimation of Sound Source Localization |
| CN104199067A (en) * | 2014-09-23 | 2014-12-10 | 南京大学 | Global navigation satellite system (GNSS) receiver fuzz-free processing method under multipath environment |
| CN105323772A (en) * | 2015-09-23 | 2016-02-10 | 浙江大学 | Self-localization method of sensor network node based on smartphone |
| CN106851011A (en) * | 2017-03-07 | 2017-06-13 | 浙江大学 | A kind of DOA estimate network system realization based on smart mobile phone acoustic array |
| CN106879068A (en) * | 2017-01-26 | 2017-06-20 | 浙江大学 | The arrival time method of estimation of signal under a kind of strong multi-path environment |
| CN109212481A (en) * | 2017-07-04 | 2019-01-15 | 北京航天长峰科技工业集团有限公司 | A method of auditory localization is carried out using microphone array |
| CN109239667A (en) * | 2018-10-26 | 2019-01-18 | 深圳市友杰智新科技有限公司 | A kind of sound localization method based on two-microphone array |
-
2019
- 2019-02-27 CN CN201910146630.2A patent/CN109959893A/en active Pending
Patent Citations (8)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102707262A (en) * | 2012-06-20 | 2012-10-03 | 太仓博天网络科技有限公司 | Sound localization system based on microphone array |
| US20140241549A1 (en) * | 2013-02-22 | 2014-08-28 | Texas Instruments Incorporated | Robust Estimation of Sound Source Localization |
| CN104199067A (en) * | 2014-09-23 | 2014-12-10 | 南京大学 | Global navigation satellite system (GNSS) receiver fuzz-free processing method under multipath environment |
| CN105323772A (en) * | 2015-09-23 | 2016-02-10 | 浙江大学 | Self-localization method of sensor network node based on smartphone |
| CN106879068A (en) * | 2017-01-26 | 2017-06-20 | 浙江大学 | The arrival time method of estimation of signal under a kind of strong multi-path environment |
| CN106851011A (en) * | 2017-03-07 | 2017-06-13 | 浙江大学 | A kind of DOA estimate network system realization based on smart mobile phone acoustic array |
| CN109212481A (en) * | 2017-07-04 | 2019-01-15 | 北京航天长峰科技工业集团有限公司 | A method of auditory localization is carried out using microphone array |
| CN109239667A (en) * | 2018-10-26 | 2019-01-18 | 深圳市友杰智新科技有限公司 | A kind of sound localization method based on two-microphone array |
Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN110376551A (en) * | 2019-07-04 | 2019-10-25 | 浙江大学 | A kind of TDOA localization method based on the distribution of acoustical signal time-frequency combination |
| CN110376551B (en) * | 2019-07-04 | 2021-05-04 | 浙江大学 | TDOA (time difference of arrival) positioning method based on acoustic signal time-frequency joint distribution |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| US9209909B2 (en) | Acoustic position-determination system | |
| CN101762806B (en) | Sound source locating method and apparatus thereof | |
| CN102324237B (en) | Microphone-array speech-beam forming method as well as speech-signal processing device and system | |
| Lazik et al. | Ultrasonic time synchronization and ranging on smartphones | |
| CN103544959A (en) | Verbal system and method based on voice enhancement of wireless locating microphone array | |
| CN203178481U (en) | Locating system based on intelligent mobile device | |
| CN106814360B (en) | A kind of multibeam sounding system based on linear FM signal | |
| CN107656244A (en) | Based on the critical indoor locating system and method for listening domain ultrasonic wave reaching time-difference | |
| CN104101863A (en) | Locating system based on intelligent mobile device and locating method | |
| Apolinário et al. | A data-selective LS solution to TDOA-based source localization | |
| CN110988799A (en) | High-precision positioning system and method for moving object in tunnel based on ultrasonic waves | |
| CN105607042A (en) | Method for locating sound source through microphone array time delay estimation | |
| Misra et al. | Acoustical ranging techniques in embedded wireless sensor networked devices | |
| Misra et al. | Improving the coverage range of ultrasound-based localization systems | |
| CN109959893A (en) | A kind of acoustical signal angle estimating method based on Beidou time service and microphone array | |
| Nishimura et al. | A proposal on direction estimation between devices using acoustic waves | |
| CN111551180B (en) | Smart phone indoor positioning system and method capable of identifying LOS/NLOS acoustic signals | |
| Hioka et al. | Estimation of direct-to-reverberation energy ratio based on isotropic and homogeneous propagation model | |
| CN108594284B (en) | TDOA (time difference of arrival) positioning performance detection method and system | |
| Li et al. | SAILoc: A novel acoustic single array system for indoor localization | |
| Nakamura et al. | Indoor localization method for a microphone using a single speaker | |
| Liu et al. | A positive and negative HFM for speed measurement | |
| KR102265743B1 (en) | Position measurement system, sound signal generation apparatus, and position measurement terminal | |
| CN111157949A (en) | Voice recognition and sound source positioning method | |
| Moutinho et al. | Indoor global localisation in anchor-based systems using audio signals |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| PB01 | Publication | ||
| PB01 | Publication | ||
| SE01 | Entry into force of request for substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| RJ01 | Rejection of invention patent application after publication |
Application publication date: 20190702 |
|
| RJ01 | Rejection of invention patent application after publication |