CN111883161A - Method and device for audio acquisition and position identification - Google Patents
Method and device for audio acquisition and position identification Download PDFInfo
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- G—PHYSICS
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- 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
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- 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
- G01S5/00—Position-fixing by co-ordinating two or more direction or position line determinations; Position-fixing by co-ordinating two or more distance determinations
- G01S5/18—Position-fixing by co-ordinating two or more direction or position line determinations; Position-fixing by co-ordinating two or more distance determinations using ultrasonic, sonic, or infrasonic waves
- G01S5/22—Position of source determined by co-ordinating a plurality of position lines defined by path-difference measurements
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- 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
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- 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
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- G10L21/0208—Noise filtering
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- G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
- G10L2021/02166—Microphone arrays; Beamforming
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Abstract
The invention discloses a method and a device for audio acquisition and position identification. According to the method, the sound source is positioned according to the audio signals collected by the microphone array, the directional pickup beams are formed according to the sound source positioning result to directionally collect the audio signals, the directionally collected audio signals are subjected to voice enhancement to obtain pure audio signals, the audio signals can be accurately collected in a complex noise environment, the directional pickup beams can suppress noise outside the beams, the quality of the collected audio signals is improved, and the subsequent audio signal identification effect is further ensured.
Description
Technical Field
The invention relates to the technical field of audio acquisition and positioning, in particular to a method and a device for audio acquisition and position identification.
Background
With the development of high-speed application of artificial intelligence technology, more and more voice positioning and recognition technology around people is applied, and like an intelligent counter of a bank, a voice recognition module is generally used for completing the handling of some intelligent services. The quality of the collected audio signal plays a crucial role in subsequent speech recognition.
For example, chinese patent document CN201320497411.7 discloses "a voice collecting device and a television system with voice collecting function", which includes at least two microphones arranged in an array, a base on which the microphones are mounted, and a wireless communication unit, where the microphones are separately arranged at positions on the base, and the wireless communication unit transmits a voice signal of the voice collecting device to a television terminal. The voice collecting device cannot position the sound source, and reduces the noise influence of the sound generating equipment by increasing the distance between the voice collecting device and the sound generating equipment.
Disclosure of Invention
The invention mainly solves the technical problems that the original audio acquisition device can not position the sound source and has poor quality of the acquired audio signal; the method and the device for audio acquisition and position recognition are provided, the audio signals can be accurately acquired in a complex noise environment by positioning a sound source and forming a directional pickup beam to directionally acquire the audio signals, and the directional pickup beam can suppress noise except the beam, so that the quality of the acquired audio signals is improved, and further the subsequent audio signal recognition effect is ensured.
The technical problem of the invention is mainly solved by the following technical scheme: the invention discloses a method for audio acquisition and position identification, which comprises the following steps:
step 1, collecting and outputting a plurality of paths of audio signals by using a microphone array;
step 2, positioning a sound source by using the audio signals collected by the microphone array in the step 1;
and 4, performing voice enhancement on the audio signal acquired in the step 3, and outputting the audio signal as a final result.
After the device enters an audio acquisition working mode, sound source positioning is carried out according to the multi-channel audio signals acquired by the microphone array, directional pickup beams are formed according to the sound source positioning result to carry out directional audio signal acquisition, voice enhancement is carried out on the directionally acquired audio signals, and pure audio signals are acquired and output. Through the positioning to the sound source and the directional collection audio signal of directional pickup beam formation, can be in the noise environment of complicacy accurate gather audio signal, and directional pickup beam can restrain the noise beyond the beam, promotes the quality of the audio signal who gathers, and then guarantees subsequent audio signal recognition effect. And the voice enhancement is carried out on the audio signals collected directionally, so that the quality of the collected audio signals is further improved, and the audio recognition effect is improved.
Preferably, the specific method for sound source localization in step 1 is as follows:
the sound source localization is determined from the position of each microphone in the microphone array and the time difference of the received audio signals.
The sound source positioning method can adopt a TDOA method, the requirement on a network is low, and the positioning precision is high.
Preferably, the method further comprises a step A between the step 1 and the step 2: and (3) carrying out voice interaction detection on the audio signal, judging whether a voice interaction signal appears, if so, executing the step (2), and if not, repeating the step (1).
The invention judges whether the device enters the working mode of audio acquisition or not through voice interaction detection, and further can reduce the calculated amount of data processing of the device, thereby reducing energy loss.
Preferably, the step a specifically includes:
a1, selecting one path of audio signal firstly acquired by a microphone array;
a2, screening a sound production starting point and a sound production ending point of a sound source in the audio signal;
step A3, judging whether a voice interaction signal exists in the signals between the sound production starting point and the sound production ending point, if so, executing step 2, and if not, repeating step 1.
The strength of the audio signal is gradually attenuated along with the extension of the distance in the air, and the shorter the propagation path of the audio signal reaches the microphone array, the shorter the time is, therefore, the strength of the audio signal which is firstly acquired by the microphone array is the strongest one of the multiple audio signals acquired by the microphone array, and the audio signal is selected to judge whether the voice interaction signal exists or not, so that the result is more accurate. The voice starting point and the voice ending point of the sound source in the audio signal are screened out, whether a voice interaction signal exists in the signal between the voice starting point and the voice ending point or not is judged, the judged audio signal interval is shortened, the accuracy of the result is improved, and the data processing amount in the judging process is reduced.
Preferably, the speech enhancement in step 4 specifically includes: and carrying out noise reduction, echo cancellation and signal amplification on the audio signals directionally collected through the directional pickup wave beams.
And the audio signals collected directionally are subjected to noise reduction, echo cancellation and signal amplification, so that the quality of the collected audio signals is further improved, and the audio recognition effect is improved.
The device for audio acquisition and position identification comprises a microphone array and an audio acquisition and processing terminal, wherein the audio acquisition and processing terminal comprises an ADC (analog-to-digital converter) data acquisition module and a processing module, the signal output end of the microphone array is connected with the signal input end of the ADC data acquisition module, the signal output end of the ADC data acquisition module is connected with the signal input end of the processing module, the microphone array is used for acquiring audio signals, the ADC data acquisition module is used for receiving and processing the audio signals transmitted by the microphone array, and the processing module is used for echo cancellation, noise reduction, voice enhancement and sound source positioning processing.
The processing module is used for analyzing the digital signals output by the ADC data acquisition module and calculating and processing echo cancellation, noise reduction, voice enhancement and sound source positioning. The processing module is internally integrated with echo elimination, noise reduction, voice enhancement, sound source positioning and other related algorithms, the problems that the traditional voice acquisition environment is high in noise, the call has echoes, the sound source positioning is inaccurate or cannot be positioned and the like are solved, meanwhile, the multiple algorithms are concentrated in the same processor, the size of the whole device is reduced, and the assembly is facilitated.
As preferred, the microphone array include back shroud, front shroud, a plurality of microphone and PCB board, front shroud and back shroud fixed connection constitute the cavity, the front shroud is equipped with the mounting hole, the microphone sets up in the mounting hole, microphone and PCB board electric connection, the PCB board sets up in the cavity, be equipped with the rubber sleeve on the microphone.
The rubber sleeve can prevent dust from entering the microphone, play a role in preventing wind and reducing noise, and improve the quality of the collected audio signal.
As preferred, the microphone array still include rubber sleeve deformation adjusting device, rubber sleeve deformation adjusting device includes main control unit, spring, clamp plate and power module, the clamp plate sets up between front shroud and rubber sleeve, spring and power module set up in the front shroud, be equipped with the spring through-hole on the face that front shroud and clamp plate contacted, the one end and the clamp plate of spring link to each other, the other end and the front shroud of spring link to each other, the spring links to each other with the power module electrical property, main control module set up in on the PCB board, main control unit links to each other with power module.
The rubber sleeve is installed in every microphone outside, and the rubber sleeve is in between microphone and the front shroud, and the rubber sleeve has certain thickness, under the exogenic action, has certain degree deformation, warp too much and arouse resonance easily, warp too little and can lose due effect. In order to enable the rubber sleeve to be in a proper deformation range, the deformation of the spring can be changed by adjusting the current flowing through the spring (the spring has initial deformation during initial installation), so that the pressure exerted on the rubber sleeve by the pressure plate is adjusted, and the purpose of enabling the rubber sleeve to be in the proper deformation range is achieved.
Preferably, the rubber sleeve deformation adjusting device further comprises a plurality of piezoelectric film sensors respectively corresponding to the rubber sleeves, the piezoelectric film sensors are arranged between the pressing plate and the rubber sleeves, and the piezoelectric film sensors are connected with the main control unit.
Whether the pressure exerted on the rubber sleeve by the pressing plate is within the proper deformation range of the rubber sleeve or not can be monitored in real time by reading the data of the film pressure sensor, and the pressure is prevented from exceeding the range when being adjusted.
Preferably, a sound insulation sponge for reducing noise is arranged in the cavity.
And the sound insulation sponge is adopted for physical noise reduction, so that the noise pollution of other areas except the area in front of the microphone to the microphone collected audio signals is weakened.
The invention has the beneficial effects that: 1) the audio signals can be accurately collected in a complex noise environment by positioning the sound source and forming a directional pickup beam to directionally collect the audio signals, and the directional pickup beam can suppress noise except the beam, so that the quality of the collected audio signals is improved, and the subsequent audio signal identification effect is further ensured; 2) whether the device enters an audio acquisition working mode is judged through voice interaction detection, and then the calculated amount of data processing of the device can be reduced, so that the energy loss is reduced; 3) the size of the current flowing through the spring is adjusted to change the deformation size of the spring (when the spring is initially installed, the spring has initial deformation), so that the pressure exerted on the rubber sleeve by the adjusting pressure plate is adjusted, the purpose that the rubber sleeve is in a proper deformation range is achieved, the rubber sleeve is prevented from deforming too much to cause resonance easily, and the deformation is too little to lose due effect.
Drawings
FIG. 1 is a flow chart of the method of the present invention.
FIG. 2 is a block diagram showing a structure of the apparatus of the present invention.
Fig. 3 is a schematic diagram of a front view of a microphone array according to the present invention.
Fig. 4 is a cross-sectional view of a side view of a microphone array of the present invention.
Fig. 5 is a cross-sectional view of an elevation structure of a microphone array of the present invention.
In the figure, the microphone array 1, the microphone array 2, the audio acquisition processing terminal 21, the ADC data acquisition module 22, the ARM/DSP processor 23, the DDR memory 24, the NAND flash memory 25, the serial port/USB communication 3, the PCB 4, the front cover plate 5, the rubber sleeve 6, the microphone 7, the main control unit 8, the piezoelectric film sensor 9, the pressing plate 10, the power supply module 11, the spring 12, the rear cover plate 13 and the sound insulation sponge 13 are arranged.
Detailed Description
The technical scheme of the invention is further specifically described by the following embodiments and the accompanying drawings.
Example (b): a method for audio acquisition and location identification according to this embodiment, as shown in fig. 1, includes the following steps:
step 1, collecting and outputting a plurality of paths of audio signals by using a microphone array;
step 2, carrying out voice interaction detection on the audio signal, judging whether a voice interaction signal appears, if so, executing step 3, and if not, repeating step 1;
and 5, performing voice enhancement on the audio signal acquired in the step 4, and outputting the audio signal as a final result.
The invention judges whether the device enters the working mode of audio acquisition or not through voice interaction detection, and further can reduce the calculated amount of ARM/DSP processor data processing, thereby reducing energy loss. After the device enters an audio acquisition working mode, sound source positioning is carried out according to the multi-channel audio signals acquired by the microphone array, directional pickup beams are formed according to the sound source positioning result to carry out directional audio signal acquisition, voice enhancement is carried out on the directionally acquired audio signals, and pure audio signals are acquired and output. Through the positioning to the sound source and the directional collection audio signal of directional pickup beam formation, can be in the noise environment of complicacy accurate gather audio signal, and directional pickup beam can restrain the noise beyond the beam, promotes the quality of the audio signal who gathers, and then guarantees subsequent audio signal recognition effect. And the voice enhancement is carried out on the audio signals collected directionally, so that the quality of the collected audio signals is further improved, and the audio recognition effect is improved.
The step 2 specifically comprises the following steps:
step 21, selecting one path of audio signal firstly acquired by a microphone array;
step 22, screening out a sound production starting point and a sound production ending point of a sound source in the audio signal;
and step 23, judging whether a voice interaction signal exists in the signals between the sound production starting point and the sound production ending point, if so, executing the step 3, and if not, repeating the step 1.
The strength of the audio signal is gradually attenuated along with the extension of the distance in the air, and the shorter the propagation path of the audio signal reaches the microphone array, the shorter the time is, therefore, the strength of the audio signal which is firstly acquired by the microphone array is the strongest one of the multiple audio signals acquired by the microphone array, and the audio signal is selected to judge whether the voice interaction signal exists or not, so that the result is more accurate. The voice starting point and the voice ending point of the sound source in the audio signal are screened out, whether a voice interaction signal exists in the signal between the voice starting point and the voice ending point or not is judged, the judged audio signal interval is shortened, the accuracy of the result is improved, and the data processing amount in the judging process is reduced.
The concrete method for positioning the sound source in the step 3 comprises the following steps: the sound source localization is determined from the position of each microphone in the microphone array and the time difference of the received audio signals. The sound source positioning method can adopt a TDOA method, the requirement on a network is low, and the positioning precision is high.
The speech enhancement in step 4 specifically comprises: and carrying out noise reduction, echo cancellation and signal amplification on the audio signals directionally collected through the directional pickup wave beams. And the audio signals collected directionally are subjected to noise reduction, echo cancellation and signal amplification, so that the quality of the collected audio signals is further improved, and the audio recognition effect is improved.
The device for audio acquisition and position identification of the present embodiment, as shown in fig. 2 to 5, includes a microphone array 1 and an audio acquisition processing terminal 2. The audio acquisition processing terminal comprises an ADC data acquisition module 21 and a processing module, wherein the processing module comprises an ARM/DSP processor 22, a DDR memory 23 and a NAND flash memory 24.
The microphone array is used for collecting audio signals in the environment, and the signal output end of the microphone array is electrically connected with the signal input end of the ADC data collection module.
The ADC data acquisition module is used for receiving the audio signals transmitted by the microphone array, converting the audio signals from analog signals into digital signals and transmitting the digital signals to the ARM/DSP processor, and the signal output end of the ADC data acquisition module is connected with the signal input end of the ARM/DSP processor.
The ARM/DSP processor is used for analyzing the digital signals output by the ADC data acquisition module and calculating and processing echo cancellation, noise reduction, voice enhancement and sound source positioning. The ARM/DSP processor is internally integrated with echo elimination, noise reduction, voice enhancement, sound source positioning and other related algorithms, the problems that the traditional voice input environment is high in noise, echoes exist in conversation, the sound source positioning is inaccurate or cannot be positioned are solved, meanwhile, the multiple algorithms are concentrated in the same processor, the size of the whole device is reduced, and the assembly is convenient. The ARM/DSP processor communicates with an upper computer through serial/USB communication 25, and the ARM/DSP processor communicates with the DDR memory and the NAND flash memory in a bidirectional mode.
The DDR memory and the NAND flash memory are used for storing data information, wherein the DDR memory is a memory and has high access speed, so that the device runs more smoothly; the NAND flash memory has the advantages of high programming speed, short erasing time and no need of any bonding logic, and can directly access instructions, addresses and data from a processor.
The microphone array comprises a rear cover plate 12, a front cover plate 4, 4 microphones 6, a PCB 3 and sound insulation sponge 13, wherein the front cover plate is in a convex shape, the rear cover plate is in a U shape, and the sound insulation sponge is in a concave shape. The front cover plate and the rear cover plate are fixedly connected through screws to form a cavity, mounting holes for placing microphones are formed in the front cover plate, one microphone is placed in one mounting hole, and 4 mounting holes are linearly and equidistantly arranged. The sponge that gives sound insulation is placed in the cavity, and the bottom side and the left and right sides of sponge that gives sound insulation contact with the back shroud, and the top side and the recess of sponge that give sound insulation cooperate the contact with the front shroud. The PCB board is placed in the recess of the sponge that gives sound insulation, and PCB board and microphone electric connection. And the sound insulation sponge is adopted for physical noise reduction, so that the noise pollution of other areas except the area in front of the microphone to the microphone collected audio signals is weakened.
The mounting hole is Y-shaped, and the side with the large opening of the mounting hole faces the inside of the cavity. The side with the large opening of the mounting hole faces the inside of the cavity, namely the side with the large opening of the mounting hole is used as a mounting hole, so that the mounting hole plays a role in guiding in the mounting process, the microphone does not need to be accurately aligned, and the mounting of the front cover plate is facilitated.
The rubber sleeve 5 is sleeved on the microphone to prevent dust from entering the microphone, so that the wind prevention and noise reduction effects are achieved, and the quality of the collected audio signal is improved.
The microphone array further comprises a rubber sleeve deformation adjusting device, and the rubber sleeve deformation adjusting device comprises a main control unit 7, 4 annular pressing plates 9, a piezoelectric film sensor 8, a power supply module 10 and a plurality of springs 11. The clamp plate sets up between front shroud and rubber sleeve, the interior anchor ring (the one side that contacts with the rubber sleeve) bonding of every clamp plate is equipped with piezoelectric film sensor, the outer anchor ring (the one side that contacts with the front shroud) 4 springs of every clamp plate, mutual interval 90 between 4 springs, the one end of spring is connected with the outer anchor ring of clamp plate, the other end of spring passes through the spring through-hole of seting up on the front shroud and the inner wall fixed connection of front shroud, the spring is located the inside of front shroud, the spring and the inside power module electric connection that is located the front shroud, power module and piezoelectric film sensor all link to each other with the main control unit communication that sets up on the PCB board.
The rubber sleeve is installed in every microphone outside, and the rubber sleeve is in between microphone and the front shroud, and the rubber sleeve has certain thickness, under the exogenic action, has certain degree deformation, warp too much and arouse resonance easily, warp too little and can lose due effect. In order to enable the rubber sleeve to be in a proper deformation range, the main control unit can know the pressure applied to the rubber sleeve by the pressing plate at the current stage by reading the data of the film pressure sensor, if the pressure is not in the proper deformation range of the rubber sleeve, the main control unit sends an adjusting control instruction to the power module, the deformation of the spring is changed by adjusting the size of the current flowing through the spring (when the spring is initially installed, the spring has initial deformation), the pressure applied to the rubber sleeve by the pressing plate is adjusted, and the purpose that the rubber sleeve is in the proper deformation range is achieved.
Claims (10)
1. A method of audio acquisition and location identification, comprising the steps of:
step 1, collecting and outputting a plurality of paths of audio signals by using a microphone array;
step 2, positioning a sound source by using the audio signals collected by the microphone array in the step 1;
step 3, forming a directional pickup beam according to the sound source positioning result in the step 2, and directionally acquiring audio signals;
and 4, performing voice enhancement on the audio signal acquired in the step 3, and outputting the audio signal as a final result.
2. The method of claim 1, wherein the sound source location in step 2 is determined by:
the sound source localization is determined from the position of each microphone in the microphone array and the time difference of the received audio signals.
3. The method of claim 1, further comprising between step 1 and step 2 the steps of: and (3) carrying out voice interaction detection on the audio signal, judging whether a voice interaction signal appears, if so, executing the step (2), and if not, repeating the step (1).
4. The method of claim 3, wherein the step A specifically comprises:
a1, selecting one path of audio signal firstly acquired by a microphone array;
a2, screening a sound production starting point and a sound production ending point of a sound source in the audio signal;
step A3, judging whether a voice interaction signal exists in the signals between the sound production starting point and the sound production ending point, if so, executing step 2, and if not, repeating step 1.
5. The method of claim 1, wherein the speech enhancement in step 4 is specifically: and carrying out noise reduction, echo cancellation and signal amplification on the audio signals directionally collected through the directional pickup wave beams.
6. An audio collection and position recognition device, suitable for the method of claim 1 ~ 5 for an audio collection and position recognition, its characterized in that includes microphone array and audio collection processing terminal, audio collection processing terminal includes ADC data acquisition module and processing module, the signal output part of microphone array is connected with ADC data acquisition module's signal input part, ADC data acquisition module's signal output part is connected with processing module's signal input part, the microphone array is used for gathering audio signal, ADC data acquisition module is used for receiving the audio signal of processing microphone array transmission, processing module is used for echo cancellation, falls and makes an uproar, speech enhancement and sound source location processing.
7. The device of claim 6, wherein the microphone array comprises a rear cover plate, a front cover plate, a plurality of microphones and a PCB, the front cover plate and the rear cover plate are fixedly connected to form a cavity, the front cover plate is provided with a mounting hole, the microphones are disposed in the mounting hole, the microphones are electrically connected to the PCB, the PCB is disposed in the cavity, and the microphones are provided with rubber sleeves.
8. The device of claim 7, wherein the microphone array further comprises a rubber sleeve deformation adjusting device, the rubber sleeve deformation adjusting device comprises a main control unit, a spring, a pressing plate and a power module, the pressing plate is arranged between the front cover plate and the rubber sleeve, the spring and the power module are arranged in the front cover plate, a spring through hole is formed in the surface of the front cover plate, which is in contact with the pressing plate, one end of the spring is connected with the pressing plate, the other end of the spring is connected with the front cover plate, the spring is electrically connected with the power module, the main control module is arranged on the PCB, and the main control unit is connected with the power module.
9. The device for audio collection and position recognition according to claim 8, wherein the rubber sleeve deformation adjusting device further comprises a plurality of piezoelectric film sensors respectively corresponding to the rubber sleeves, the piezoelectric film sensors are disposed between the pressing plate and the rubber sleeves, and the piezoelectric film sensors are connected to the main control unit.
10. The device of claim 6, wherein a sound-proof sponge is disposed in the cavity for reducing noise.
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Cited By (3)
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CN112565974A (en) * | 2020-11-30 | 2021-03-26 | 神思电子技术股份有限公司 | Method for solving bidirectional separation pickup |
CN113794963A (en) * | 2021-09-14 | 2021-12-14 | 深圳大学 | Speech enhancement system based on low-cost wearable sensor |
CN114325214A (en) * | 2021-11-18 | 2022-04-12 | 国网辽宁省电力有限公司电力科学研究院 | Electric power online monitoring method based on microphone array sound source positioning technology |
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