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    • 专利摘要:
    digital speaker system with channel equalization and beam control functionalities and related methods wherein the method comprises the steps of: (1) digital format conversion, to convert the original signals to digital signals based on pcm encoding; (2) channel equalization processing; (3) control speaker formation; (4) performing multi-bit (sigma-delta) modulation; (5) thermometer code conversion, to convert low-bit encoded pcm signals with a bit width of m into unary code vectors of a digital power amplifier and a transducer load corresponding to 2m transmission channels; (6) dynamic mismatch shaping processing to reorder the thermometer encoded vectors; and (7) extracting channel information and sending it to the digital amplifier and directing sound load.
    公开号:BR112014009896B1
    申请号:R112014009896-4
    申请日:2011-12-28
    公开日:2021-06-22
    发明作者:Dengyong Ma
    申请人:Suzhou Sonavox Electronics Co., Ltd.；
    IPC主号:
    专利说明:

    Invention sector
    The present invention relates to a method and device for channel equalization and beam control, in particular to a method and device for channel equalization and beam control for a digital speaker system. state of the art
    With the rapid development of large-scale integrated circuit and digital technology, the inherent defects of the conventional analog speaker system are becoming increasingly evident in power dissipation, volume and weight, as well as in transport, storage and processing. of signs and the like. In order to overcome these defects, speaker system research and development is gradually turning to low power dissipation, small profile, digitization and integration. As the emergence of class AD digital power amplifier, based on PWM modulation, the course of digitization of the speaker system has advanced to the power amplifier part, however, high quality inductors and high-volume, high-priced capacitors they are still needed so that the post-stage digital power amplifier circuit can passively simulate low-pass filtering to eliminate high-frequency carrier components in order to demodulate the original analog signals.
    In order to increase the volume and cost of digital power amplifiers and get more integration US patents US 20060049889A1, US20090161880A1 disclose digital speaker system based on PWM modulation and BD class power amplification technology. However, there are two significant disadvantages in digital speaker systems based on PWM modulation: (1) the coding scheme based on PWM modulation has inherent non-linear defects due to its modulation structure, causing the encoded signals to generate distortion components non-linear in the desired band, whereas, if more linearization means are employed to improve the result, the realization difficulty and complexity of the modulation form will rise sharply; (2) in view of the difficulty of realizing hardware, the oversampling rate of PWM modulation is low, usually in the range of 200 KHz ~ 400 KHz, making SNR (Signal-to-Noise Ratio) of the encoded signals can no longer be increased due to oversampling rate limitation.
    Considering the defects of non-linear distortion and low oversampling rate of the PWM modulation technique in the implementation of the digital speaker system, with the full digital demand of the entire signal transmission chain, the China patent CN 101803401A discloses a digital speaker system based on ∑-Δ multi-bit modulation. In such a system, high-bit PCM code is converted to unary code vector as a control vector to control the on-off action of the speaker arrangement, by multi-bit ∑-Δ modulation and thermometer coding techniques , and the high-order harmonic components of the synthetic spatial domain signals resulting from the frequency response difference between the matrix elements are eliminated by the incompatibility shaping dynamics technique; although the system disclosed by the patent performs all the digitization of the entire signal transmission link and reduces the total harmonic distortion rate of the synthetic spatial domain of signals by the incompatibility dynamic shaping technique, however, the incompatibility dynamic shaping technique does not have equalization effect on the frequency response fluctuation in the audio band of the channel, so a large deviation between the system restore signal spectrum and the sound source is caused by the frequency response fluctuation in the band of each channel, so there is a big difference between the sound field of the restoration and the real sound field, making the digital reproduction system unable to reproduce the real sound field effect of the original sound source. Additionally, this fluctuation of frequency response in the band of each channel also causes low stability and slower convergence rate of the various self-adaptive beamforming algorithms, thus causing the robustness of the self-adaptive beamforming algorithms to be poor.
    Now, the beam steering method based on channel delay regulation described in Chinese patent CN 101803401A is a simple beamforming method, which only regulates the phase information of the transmission signals of each matrix channel, without considering the regulation of the magnitude of the transmission signals of each channel. The beam control capability provided in the method is weak and certain beam steering capability is provided only in the environment adjacent to the free field in the method, in some cases, based on such a delay control method, it is not possible to perform the control multi-beam steering, when it is necessary for the digital system to generate multiple directional beams. Also, in practical application, there are usually many spread limits, this makes the transmitted signals contain a large amount of multipath scattered signals in addition to direct sound. In this reverberant environment of obvious multipath dispersion, the best beam directional control cannot be achieved just by relying on the channel delay control direction method. Consequently, considering the beam directional control problem of the digital loudspeaker array in a reverberant environment, it is necessary to look for a complex beamforming method that has the anti-reverberation capability, to simultaneously regulate the amplitude and phase of the signals transmission of each channel, thus achieving the desired sound field control effect. Currently, almost all digital matrix systems based on multi-bit ∑-Δ modulation rely on the conformation mismatch technique to eliminate the frequency response difference between multiple channels, however, such method of correction for the response difference channel frequency only adapts to correction of a small frequency response deviation, and the ability to correct phase deviation is quite weak. Furthermore, said shaping technique has no equalizing effect on the frequency response variation in band of each channel, while the frequency response variation of these channels would bring to timbre the variation of the sound field restoration, so it is difficult ensure full recovery of the sound field. The beam control method employed in conventional digital speaker arrangements is a simple channel delay control method and such method only adapts the ideal free sound field environment, the method will not be suitable when many multi-path differences arise in the sound field due to reflection or scattering. In some applications, the delay control-based method cannot achieve multi-beam sound field control effect, when it is necessary for arrays to generate multiple directional beams.
    Considering the shortcomings of existing digital speaker array systems based on ∑-Δ multi-bit modulation in channel equalization and beam control, a more effective method of channel equalization and beam control is needed to satisfy the demand for application of a digital speaker arrangement system based on ∑-Δ modulation in flatness of frequency bands and beam directivity, requiring a digital speaker arrangement system device having channel equalization and control functionalities. of beam. Invention Summary
    In order to overcome the defects of the digital speaker system in channel equalization, the present invention provides a method of channel equalization and beam control for a digital speaker array system as well as a speaker system device. digital speaker with channel equalization and beam control features.
    For the above-mentioned purpose, the invention provides a method of channel equalization and beam control for a digital speaker array system, comprising the following steps: (1) digital format conversion, to convert the signals to digital signals based on PCM encoding; (2) The realization of channel equalization; (3) Control beam formation; (4) The multi-bit modulation realization ∑-Δ; (5) The realization of thermometer code conversion, to convert the PCM low-bit encoded signals with bit width M into digital power amplifier unary code vectors transducer load corresponding to 2M transmission channels; (6) perform dynamic mismatch shaping processing, to reorder the thermometer encoded vectors, and (7) Extract channel information, to send to digital power amplifier and drive sound load.
    Furthermore, the digital format conversion in step (1) can be directed to analog and digital signals. For analog signals, the signals must be converted to digital signals based on PCM encoding by the analog-to-digital conversion converter, before being converted to PCM encoded signals that meet the parameter requirements according to bit width and rate demand parameter designated sampling devices. For digital signals, the signals are converted into PCM encoded signals that satisfy the parameter requirements according to the designated bit width and sample rate parameter demand.
    Preferably, for processing the equalization channel, in step (2), the equalizer parameters can be achieved according to the measurement method provided, where the number of elements is N, the number of measurement points in the location desired is H, and the white noise signal emission elements s(t), the response pulse hi,j for the channel element of the desired measurement point location can be calculated by obtaining received signals r(t ) at the measured point, where i represents the index number of element No. i, and j represents the index number of measuring point no. j in the desired region. Since all impulse responses
    of element no. i for all measurement points have been calculated, then the average impulse response * of element no. i of the desired region can be obtained by a weighted fit method, where Wj represents the weighted vector of the element of frequency response no. i at measuring point n° j. Then, the inverse filter response hi'1 of the average impulse response hi can be calculated according to the inverse filter estimation algorithm. Finally, the result of convoluting the average impulse response h1 of the first element to the desired location and the inverse filter response i- i selected as the reference vector hr = h1 * h1'1 , then the inverse filter response hi'1 (2 < i < N) of the residual element channels is compensated by the compensation factor setting of hc, the result of the convolution hi,r = hi * hj,c'1 of the compensation result hi,c'1 = hc * hi'1 and the average impulse response hi is completely equal to the reference vector hr, thus obtaining the equalizer vector response as follows:

    Further, for the beamforming control in step (3), the weighting coefficient of the beamformer channel can be calculated by a normal beamforming method.
    Since the number of matrix elements is N, their spatial domain direction vector is:
    The desired beam configuration in the special domain is:
    As long as the weighted coefficient of the array vector to be calculated is
    , then the formula for calculating the weighting coefficient of the array can be obtained by the least squares criterion, as follows:
    The transmission signals of each channel are regulated in magnitude and phase, using the weighted average of the array vector, thus orienting the spatial domain that emits the array's acoustic beam to the desired region.
    In addition, the multi-bit modulation process ∑-Δ in step (4) is as follows: firstly the PCM-bit high codes after equalization processing are subjected to interpolation filtering by an interpolation filter in terms of the so-called oversampling factor, to obtain oversampled PCM encoded signals; and then the sound energy within the audio bandwidth is pushed out of the audio band by ∑-Δ modulation processing, to ensure that the system has sufficiently high SNR band. While the original high-bit PCM codes are converted to low-bit PCM by processing modulation codes ∑-Δ, and the number of PCM codes is reduced.
    Preferably, step (4) of multi-bit ∑-Δ modulation performs the noise shaping process on the output of the over-sampled signals from the interpolation filter by using various existing ∑-Δ modulation methods, such as the modulation method High-order single-stage series or multi-stage parallel modulation method (Cascade, MASH), to push the noise energy out of the band and still ensure that the system has sufficiently high SNR band.
    Further, the thermometer code conversion in step (5) is for converting low bit PCM encoded signals with a width of into unary code vectors of a digital power amplifier and corresponding transducer load to 2M transmission channels. The codes of each digit of the unary code vectors will be sent to the corresponding digital channel. Each digit code has two level states of “0” or “1” at any time, where in state “0” the transducer load will be turned off, while in state “1” the transducer load will turn on. The thermometer encoding operation is to assign the encoded information to multiple load channel transducers, thus bringing the transducer load into the signal encoding stream and obtaining the digital encoding and digital switching control of the transducer array.
    Additionally, the dynamic mismatch shaping process in step (6) is to reorganize the thermometer coded vectors, to further improve the data allocation scheme of the unary code vectors and eliminate high-order harmonic distortion components of the synthetic signals of the spatial domain arising from the frequency response difference between elements of the array.
    Furthermore, the dynamic mismatch shaping in step (6) conforms the nonlinear harmonic distortion spectrum arising from the frequency response difference between elements of the array, using several existing shaping algorithms, such as DWA (Data-Weighted Averaging) algorithms , VFMS (Vector-Feedback mismatch-shaping) and TSMS (Tree-Structure mismatch-shaping), to reduce the magnitude of harmonic distortion in the band and push power out of the high frequency section of the band, thus reducing the magnitude of harmonic distortion in-band and improving the sound quality of ∑-Δ encoded signals.
    Still, extracting channel information in step (7) refers to performing the operation of distributing coded information for each channel, and the signal processing process is as follows: first the dynamic mismatch formatter of each channel performs the dynamic mismatch shaping process to obtain reordered shaped vectors, and then a designated digital code is selected from the shaped vector digits of each channel according to certain selected extraction criteria. To ensure complete restoration of information, the number of selected digits of one channel must be different from those of other channels, and all digits of selected number orders of all channels completely contain the order of digits from 1 to
    During the course of the selection operation in extracting channel information, digit selection is usually performed by a simple rule, eg, in channel no. i, encoded information of digit no. i is selected from the shaping vectors the same. After the channel bits are selected and combined, the equalization and adjustment of the weighted beam processing in the channel elements of the multi-array are efficiently performed, thus providing an effective means of realizing the equalization and directive control of the digital arrangement.
    Preferably, the load in step (7) can be a digital speaker arrangement comprising multiple speaker units, or a speaker unit having multiple voice coils, or alternatively a digital speaker arrangement comprising a plurality of speaker units. speaker of multiple voice coils.
    The present invention also provides a digital speaker array system having channel equalization and beam control features, comprising: a sound source, which is the information to be reproduced by the system; a digital converter, which is electrically coupled to the output end of the sound source, to convert the input signals into high-bit PCM encoded signals with a bit width of N and a sampling rate of
    A channel equalizer, which is electrically coupled to the output end of the digit converter, to perform reverse filtering equalization on the frequency response of each channel to eliminate frequency response fluctuation in the channel band; a beamformer, which is electrically coupled to the output end of the channel equalizer, to control the channel beam shape in the spatial domain of the speaker and create the sound field distribution characteristics such as 3D stereo sound field, field virtual surround sound and directional sound field and the like, to achieve the purpose of playing a special sound effect; a ∑-Δ modulator, which is electrically coupled to the output end of the beamformer, to obtain oversampling interpolation filtering and ∑-Δ multi-bit code modulation, and to obtain low bit PCM encoded signals with a bit width reduced;
    A coded thermometer, which is electrically coupled to the output end of the ∑-Δ modulator, to convert the low bit PCM coded signals to unary vectors, which are equal in quantity to the system's digital channels, thus digitizing the switching control vectors of the channel; a dynamic mismatch formatter, which is electrically coupled to the output end of the thermometer encoder, to eliminate distortion of nonlinear spatial domain harmonic components arising from the frequency response difference between the elements of the array, reducing the magnitude of the harmonic distortion components in-band, and forcing the high-frequency harmonic frequency power components out-of-band and improving the sound quality of the encoded signals ∑-Δ; an extract selector, which is electrically coupled to the dynamic mismatch formatter, to extract coded information from certain digits of the conformation vectors of each channel, and control channel on/off information; a multi-channel digital amplifier, which is electrically coupled to the output end of the pull selector, to amplify the power of the coded control signals and drive the on/off action of each channel and drive the on/off action of the post-stage digital load ; a digital array load, which is electrically coupled to the output end of the multi-channel digital amplifier, to achieve the electro-acoustic conversion and conversion of the switch's digital electrical signals into analog air vibration signals.
    Furthermore, the sound source can be analog signals generated by several analog devices or digital encoded signals generated by several digital devices.
    Preferably, the digital converter can be compatible with existing digital interface formats, it can contain analog-digital converter, digital interface circuits such as USB, LAN, COM and the like, and interface protocol programs. Through the interface circuits and protocol programs, the digital speaker array system can interact and transmit information with other devices flexibly and conveniently. Meanwhile, the original input analog signals or digital sound source signals are converted to high bit PCM encoded signals with a bit width of N and sampling rate of fs by digital converter processing. In addition, the channel equalizer can perform equalization processing in terms of time-domain or frequency-domain inverse filtering response parameters and eliminate frequency response oscillation in the band of each channel, while the frequency response difference of each channel can be corrected, thus making the frequency response difference of each channel tend towards consistency.
    In addition, the beamformer performs weighted processing in the transmission of each channel's signals, using the designated weighted vectors, to regulate the magnitude and phase of their information, making the spatial domain pattern of the digital array of a complicated environment satisfying the pattern. desired demand.
    Preferably, the signal processing process of the ∑-Δ modulator is as follows: in principle, the encoded PCM signals with bitwidth N and sampling rate of fs are subjected to oversampling interpolation filtering in terms of oversampling factor to obtain the encoded PCM signals with bitwidth N and a sampling rate of osmf and then the encoded oversampling PCM signals with bitwidth N are converted to low-bit encoded PCM signals with bitwidth M (M < N), thus reducing the bit width of the PCM encoded signals.
    In addition, the ∑-Δ modulator can perform noise shaping processing on the output of oversampled signals from the interpolation filter, according to the signal processing structures of various existing ∑-Δ modulators, such as the -Δ modulator structure. single-phase high-order serial modulator or parallel multi-stage modulator structure and drive the noise energy out-of-band to ensure that the system has sufficiently high SNR band.
    Preferably, the thermometer encoder is used to convert the PCM encoded signals of low bit bit width M into unary digital code signal vector of load amplifier and transducer corresponding to 2M channels. The encoded information of each digit of the unary vector code of each channel is assigned to a corresponding digital channel to bring the transducer load to the signal coding stream, thus achieving digital coding and digital transducer load control switch.
    In addition, the dynamic mismatch formatter uses several existing shaping algorithms such as DWA (Data-Weighted Averaging), VFMS (Vector-Feedback mismatch-shaping) and TSMS (TreeStructure mismatch shaping) algorithms to shape non-linear harmonic spectrum distortion arising from the frequency response difference between the array elements, to reduce the magnitude of the harmonic distortion components in the band and push the energy to the high frequency section out of the band, reducing the magnitude of harmonic distortion and improving the quality of sound of encoded signals ∑-Δ.
    Preferably, the extraction selector acts according to a certain rule of extracting one-digit information from the formation vectors of each of the channels of 2M digital channels as the corresponding channel exit code information, to control the action to turn on/off the load of the post-stage transducer. After bit extraction and extraction selector merging operation, the response equalizer operation and channel directivity vector weighting of the various original channels is performed effectively, which ensures frequency response flatness
    Digital arrangement and controllability of beam direction.
    In addition, the multi-channel digital amplifier sends the output of the pull selector switching signals to a full-bridge power amp circuit MOSFET grid. The on/off status of the circuit and the power supply to the load can be controlled by controlling the on/ff status of the MOSFET, thus achieving the power amplification of the digital load.
    Preferably, the digital array payload may be an array containing multiple digital speaker units, or a multi-voice coil speaker unit, or alternatively be an array of speakers comprising speakers of multiple voice coils. Each digital channel of the digital payload may comprise one or more speaker units, or one or more voice coils, or alternatively comprise multiple voice coils and multiple speaker units. The digital load matrix configuration can be arranged according to the number of transducer units and practical application, so as to form various matrix configurations.
    The present invention has the following advantages over the prior art: a). The invention achieves the entire digitization of the entire signal link transmission, the entire system of the invention consists of digital devices and thus highly facilitates the design of the integrated circuit, and the invention improves the working stability of the system as well as decreases dissipation of power, volume and weight of the system. In addition, the joint digital speaker system provided by the invention can achieve data exchange with other digital media flexibly and conveniently, and can adapt to developing digitization demand better. B) . The multi-bit ∑-Δ modulation employed in the invention pushes the noise energy to the high frequency region out of the band by noise shaping, thus ensuring high SNR demand in the band. The hardware realization circuits of this modulation technique are simple and inexpensive, and have excellent immunity to parameter deviations caused in the manufacturing process of the circuit elements. c) The fully digital system of the invention has great anti-interference capability and can work stably in the complicated environment of electromagnetic interference. d) The dynamic mismatch shaping algorithm used in the invention can effectively eliminate the magnitude of non-linear harmonic distortion arising from the frequency response difference between the matrix elements and improve the sound quality of the system and therefore the system. invention has excellent immunity to frequency response drift between transducer units. e) The thermometer coding method applied in the invention can allocate corresponding unary code signals to each transducer unit, making each speaker unit (or each voice coil) work in on/off state, as such an alternative of On/off working state can avoid overload distortion phenomenon of each speaker unit (or each voice coil), thus extending the service life of each speaker unit (or each voice coil). Furthermore, the transducer can achieve greater electro-acoustic transformer efficiency and generate less heat by using the on/off way of working. f) The digital power amplifier circuit applied in the invention sends amplified switching signals to the loudspeaker and also controls the on/off action of the loudspeaker, without adding any high-volume and costly inductors and capacitors in the post circuit. -phase power amplifier for analog digital processing low-pass, thus decreasing the volume and cost of the system. In addition, for piezoelectric transducer load with capacitive characteristic, it is usually necessary to add inductor for impedance matching to increase the acoustic output power of the piezoelectric speaker, and the corresponding impedance matching effect of applying digital signals to the transducer is superior to even applying the analog signals to the end of the transducer. g) The thermometer coding scheme used in the invention causes the unary code signals allocated from each array element set to contain only part of the information from the original sound source signals, thus the sound source information cannot be fully restored simply by relying on information emitted from a single array of elements, therefore, full restoration of sound source information can only be achieved by combining the sound field spatial domain effects synthetic of all sound field sets. elements of the issuing matrix. Furthermore, the restored information obtained by combining above has spatial domain directivity and has the maximum SNR on the symmetry axis of the array, and the SNR decreases with distance from the major axis. h) The channel equalization method of the present invention can maintain the frequency response in the flat band and correct the frequency response difference between channels; this causes the system-restored sound source signal spectrum and the actual signal spectrum of the original sound source to tend to consistency, thus ensuring that the playback system actually reproduces the field effect sound of the sound source. original. Meanwhile, the smoothing of the frequency response of the band of each channel and the consistency of the frequency response in the band between the channels resulting from the method provides favorable support for the best stability, the highest convergence rate and the best robustness of the various algorithms self adaptive. i) The channel equalization method based on the data extraction selection provided in the invention can efficiently suppress the frequency response fluctuation of each channel and improve the field-restored sound quality of the digital system as well as eliminate the big difference frequency response between the channels and therefore the frequency response difference between the channels can be compensated to a large extent after multi-channel equalization processing, and only some residual deviations remain, while these residual deviations may be even more efficiently corrected by relying on the conformation mismatch algorithm, thus making the mismatch conformation algorithm's ability to eliminate some deviations can be brought into full touch. The frequency response difference of matrix elements can be efficiently corrected through equalization channel processing, thus ensuring the various matrix beam control algorithms based on coherent accumulation of matrix element channels can work efficiently . Such a digital matrix beamforming method based on data extraction selection can efficiently improve the ability of digital matrices to control the spatial sound field in complicated environment. J. The beam control method applied in the invention ensures that digital speaker arrangement has better beam directivity in the complicated environment, by combining extraction selection information, the normal beam control method can be applied in such a way. efficient in digital matrix beam control, which provides an effective implementation way for generating special sound field effects in practical environment, such as 3D stereo sound field, virtual surround sound field, and directional sound field and similar. K) In the data extraction selection method employed in the present invention, conventional channel equalization and beamforming algorithms based on PCM encoding format can be applied directly to digital matrix systems based on multi-bit modulation ∑- Δ, thus creating a bridge between channel equalization and conventional beam controller algorithms and digital matrix systems based on ∑-Δ multi-bit modulation, and ensuring that conventional algorithms can continue to play the role of channel equalization and direction of the beam effectively in matrix systems based on ∑-Δ modulation. Brief description of the dosages
    Figure 1 is a block diagram illustrating the modules that make up a digital speaker system device with channel equalization and beam control functionalities, according to the present invention;
    Figure 2 is a schematic view illustrating the measurement of channel parameters in the process of estimating equalizing channel parameters in accordance with the present invention;
    Figure 3 is a schematic view showing the channel weight vector loading in the beam control process in accordance with the present invention;
    Figure 4 is a schematic view showing the extraction rule used in extracting channel information, according to the present invention;
    Figure 5 is a graph illustrating the magnitude of the spectrum of inverting filters used in the channel equalization process, according to the present invention;
    Fig. 6 is a flowchart showing the signal processing of a fifth order CIFB modulation structure used by the ∑-Δ modulator in accordance with the present invention;
    Figure 7 is a schematic view illustrating the on/off control of the thermometer encoded vector, in accordance with the present invention;
    Figure 8 is a flowchart showing the mismatch shaping algorithm used by the dynamic mismatch formatter in accordance with the present invention.
    Figure 9 is a schematic view showing the extraction rule used by the extraction selector, according to an embodiment of the invention;
    Figure 10 is a schematic view showing the layout of the speaker array and eight elements, according to an embodiment of the invention;
    Figure 11 is a schematic view showing the local configuration of the speaker arrangement and microphone unit, according to an embodiment of the invention;
    Figure 12 is a comparison graph illustrating the magnitude spectra of the frequency response of the system before and after equalization at the location point one meter away from the matrix axis, in accordance with an embodiment of the invention ;
    Figure 13 is a graph illustrating beam patterns produced in the three predetermined directions of -60 degrees, 0 degrees and 30 degrees, in accordance with one embodiment of the invention;
    Figure 14 shows the parameter values used by the ∑-Δ modulator, according to an embodiment of the invention. Detailed description of the invention
    The present invention will be described below in conjunction with the drawings and specific embodiments of the present invention is described in more detail - digital converter of the present invention connected to the first via an audible audio signal within the width of N bits is within the high-bit range PCM encoded signal, then use channel equalization, digital audio signal for each channel reverse filtering equalization to eliminate the response of each audio channel with highs and lows frequency, eliminating the difference between the response in frequency, so Lee Abeam forming technology, the equalized signal is weighted for each channel, so that the matrix can be directed to the desired spatial direction; then use multi-bit ∑ - Δ bandwidth modulation for the high bit PCM coded signal in N bit width becomes M (<N) low bit coded PCM signal, and The thermometer approach coded through the PCM coded signal to the bit width 2 bits MM f temperature coded, forming two groups assigned to Chen unary code signal transducer elements, then through dynamic incompatibility shaping techniques assigned to each group of the element of the arrangement for the dynamic formation process unary code signal mismatch, eliminate the difference in the response of the frequency array element groups introduced higher harmonic components, reducing the total harmonic distortion of the system, improve the system level of sound quality; Finally, the decimation selection technique, the incompatibility of shaping the vector from each channel, the bit information is extracted by one bit, for the digital signal of the power amplifier channel is formed, the channel conduction charge Digital can be open or close operation, every sound field of the radiated airspace payload digital channel after the source signal reduction is superimposed within a predetermined area of space. 1, a production apparatus according to the speaker system with equalization and control functions of the digital channels of the present invention, the beam, which is the main source 1, a digitizer 2, equalizer channel 3, 4 shaper beam, ∑ ~ Δ modulator 5, encoder temperature 6 Adynamics mismatch modeler 7, decimation selector 8, digital multi-channel digital amplifier 9 and load 10 is composed of a matrix.
    Audio 1, you can use PC audio files in hard disk storage MP3 format, USB port through digital format output: you can also use MP3 player to store the audio files through analog format output, a test signal within the audio range can also be generated by the signal source, the output is of an analog format.
    Digital Converter 2, and the audio output of a link, including the input format of the input format of two digital and analog input interfaces to digital input format, using Ti's PCM2706 USB interface chip model, the PC capability the machine is stored MP3 files through the USB port, according to the type of] 6 bits, sampling rate of 44.1 KHz in real time, via I2S interface protocol is read in the Cyclone III model EP3C80F484C8 FPGA core jf; For the analog input format, using the Analog Devices a model for the AD1877 analog-to-digital converter chip, the analog audio signal in i6-bit, the encoded signals in PCM 44.1 kHz, in real time, via interface protocol I2S read on FPGA chip.
    Equalizer channel 3 is connected with the terminal of said output digital converter 2, according to the measurement method to calculate the inverse of the filter parameters for each channel, Figure 5 shows the amplitude spectrum of the reverse curve channel filter 1 -8, in accordance with the inverse filter parameters for each channel equalization processing, 16-bit, 44. 1 KHz sample rate for PCM í P if equaled ! J.
    Beam former 4, the channel equalizer is connected to the output terminal 3, according to the desired beam pattern 8 yuan calculated weight vector matrix, then the FPGA, the weight vector calculated by the multiplier unit is loaded into each element of the transmission signal channels - 16 bits after equalization, 44 1KHz sampling rate of PCM signal, thus forming with weighted multichannel PC signal direction. ∑ - Δ modulator 5, and the beamforming output terminal 4 is connected, first, the FPGA chip, the interpolation filter for over-sampling operation, 44: I K'HZ, 16-bit PCM encoded signal, press . upsampling performed three interpolation, the first stage interpolation factor 4, the sampling rate is promoted π6, 4KHZ, the second stage interpolation factor 4, the sampling rate is promoted 705.6KHZ, Chapter E: interpolation factor 2 phase, the sampling rate 1411.2 KH promoted zeta "32 times after interpolation, the original 44.1 KHZ, 16-bit PCM signal is converted to 1.41] 2 MHz, oversampled PCM signal] 6-bit, then follow the 3-bit ∑-Δ modulator way, the sample had oppose MHz, PCIV 1.4112 16 bits [PCMB encoded signal into 1.4112 MHz encoded signal> 3 bits, as shown in Figure 6, in the present embodiment, modulator ∑-Δ using five bands CIFB Cascaded Integrators with Distributed feedback) topology. modulator coefficient shown in Table 1, in order to save hardware resources and reduce the facts modern parts of the FPGA chip normally used instead of the constant multiplication viper change and Modulator parameters ∑-Δ is represented by using CSD encoding.
    Temperature Acoder 6 connected to the output terminal ∑ Δ 5 modulator will 1.4112 MHz, ∑-Δ 3-bit signal, according to the modulation coding is converted to a thermometer 1.4112 MHz, the code width 8 bits unary . 7, when the three-bit PCM encoded as "001", the conversion of the "00000001 encoding is used to control the matrix element of an open matrix transducer encoded by a thermometer, the remaining seven matrix elements are turned off; if 3 PCM bit coded as "100", the conversion of a-coded thermometer to "00001111", the coding unit to control the presentation of the transducer array 4 element open, the remaining four element MR outside; When PCM coded as three bits "111" converts the to-coded thermometer "01111111", to control the transducer 7-coding arrangement element open, leaving only one element of the arrangement is closed.
    Mismatch dynamic modeler 7, the encoder and a thermometer connected to output terminal 6, for the elimination of non-linear frequency response of harmonic distortion components caused by the difference between the array elements. Dynamic mismatch modeler 7 minus a nonlinear harmonic distortion optimization criterion, if the temperature of 8 encoding type 8 to determine the distribution encoding transducer element, shown in Figure 7, when the temperature if encoded as "OOOOUil", through the dynamic mismatch shaper be arranged in order, it will determine the transducer array element 1, 4, 5, 7 distribution code "1", the transducer array elements 2, 3, 6 distribution coded in 8 "0", and according to this Cong array transducer element distribution 1, 4, 5, 7 and will open the transducer array elements 2, 3, 6, 8, will be closed, according to this assignment of control codes to turn on the transducer assembly, so that the display will put the synthesized acoustic radiation signal contains a minimum of harmonic distortion components. In the present embodiment, the VFMS dynamic mismatch modeler algorithm used, the signal processing flow shown in Figure 8, in which the bold line represents V-vector represent thin scalars, V is the modulator input signal ∑-Δ and code W-dimensional vector after processing the coded by thermometer, the vector code V contains a state "]" and A""A"0" state, the output signal SV is a mismatch processing process after N-dimensional vector column , by incompatibility process of framing, the output vector "1" of the state and "0" state in the vector order has been adjusted, but the "1" of the state and the number "0" state remains unchanged, and once every one of the element vector controls the element of the array corresponds to a variety of on-off operation channels according to its status. Unit selection module through some selection strategy to ensure that the error introduced by the frequency response can be the difference in spectrum
    Get better shaping effect, - modulo min() represents the smallest element of selected numerical N-dimensional vectors while its negation by - scalar elements Ming modulus operation is obtained u, mtf is function shaping incompatibility, which The general form is 1 , H is the end of the present mismatch former order embodiment employed in the order 2 embodiment. expression vector output is - sv = u[L 1...1], chi iY + mif(se), where sv = sv~~y. Assuming inconsistencies representation v-dimensional matrix output sound error line array array between units, ü assume that the sum of all elements is zero, then the speaker matrix in space points anywhere each element of the matrix superimposed field of sound synthesis output obtained after signal expression is:

    Expression matrix output acoustic signals can be seen, the mff shaping function can be shaped arrangement error handling, just choose a better shaping mtf function mismatch, you can get a better range of error shaping effect. In the FPGA chip, the plastic transformation dynamic mismatch, the original ∑ - Δ harmonics present in the signal encoding pushed to high frequency band, thus increasing the level of sound quality in the audio band signal.
    Decimation selector 8, output terminal 7 is related to the dynamic mismatch shaper to shape vector in each channel for digital extraction operation, sent after the amplifier and digital load. Shown in Figure 9, each channel mismatch by a shaping process produced a $8 million extraction selector 7 vector code will be pulled in the shaping vector principle according to the first digit of the i-th channel, each channel is extracted a corresponding one yuan digital code signal, as the digital level amplifier input signal.
    Digital multi-channel amplifier 9, and having the selector sleeve 8 is connected to the output terminal. This mode, the Ti digital amplifier chip selection is a model for TAS512] digital amplifier chip, the chip response time to 100 ns magnitude, the response may be distortion of a dollar 1.4112 MHz signal flow. of the amplifier, differential input format, within the FPGA, dynamic incompatibility shaping the way the output data sent directly to the output, after passing another inverting output, forming two differential signal to a chip differential TAS5121 input terminal and, to the output amplifier, a differential output using the same format, the two differential signal is directly applied to a single transducer element of the positive and negative conductor channel.
    Display digital load] 0, and multi-channel digital amplifier connected to the output terminal 9. In this embodiment, the digital load cell produced using the Huiwei B2S model of speaker unit bandwidth, the range unit frequency from 270 Hz to 20 KHz, sensitivity (2.83 V / LNI) to 79 (IB, maximum power 2 W, nominal impedance of 8 ohms. shown in Figure 10, a load of 8 yuan digital matrix speakers, the screen 8 by the above-described speaker unit placed in a linear arrangement mode, the matrix element spacing of 4 cm, each speaker unit corresponding to a digital channel.
    In the free space, assuming the cloth speaker matrix and place the microphone unit as shown in the] 1, according to simulation methods, digital speaker system device for input frequency range 100Hz ~ 20KHz assumptions scan signal, the frequency response speaker array axis 1 meter the position of the observation systems point. Figure 12 shows the before and after the equalizer is applied, the axial position of a meter point of the frequency response curve of the amplitude spectrum system of the comparison chart, the equalizer is not applied, when the spectrum response of 2KHz amplitude ~ 20KHz frequency range system After applying equalizer; i Jane memory in a very clear decay trend, with the frequency from 2KHz to 20KHz, the system frequency response amplitude spectrum decreased from 65dB to 45dB, 20dB difference in the magnitude of the existence of The response spectrum System frequency amplitude was kept within 2KHz~20KHz frequency range of 57dB in the vicinity, showing a very flat spectrum, thus ensuring the real reduction system synthesis signal. According to the equalization results, the use of multiple channel selection bits to extract information about the synthesis mode, the equalizer can effectively inherit the channel response information to ensure smoothing the response of each of the frequency channels .
    Digital channel equalization based on the speaker matrix system can effectively eliminate the sound response of each channel within the oscillation frequency band, and correct the frequency response difference between the channels to ensure that the system has a smooth response. very flat frequency in the time domain within a desired region of space characteristics, thus ensuring the space of all channels in the spectrum of the synthesized signal can restore the original audio signal, the real spectrum, to ensure that the system is a faithful representation of the sound field of the original digital reproduction tone effects. In addition, by eliminating the audio band frequency response of each channel to ensure that the various highs and lows also adaptive spatial matrix beamforming algorithm has a faster convergence speed and better robustness.
    In free space, the array of speaker placement process is further shown in Figure il, according to the - 60 degrees, 0 degrees and 30 degrees predetermined three main lobe direction, in which the beam control matrix of simulation, a set of three beamwidth layout conditions are 20 degrees. Figure 13 shows three predetermined spatial distribution pattern of the matrix T case direction, observation of these curves can be seen, the main lobe of the matrix pointing to a predetermined direction, to achieve a desired effect beam width requirements Sidelobe amplitude difference achieved 15 dB, according to the matrix results beam control, mining extraction, multi-channel information to select the synthesis method, it is possible to inherit the beamformer effectively loaded in the amplitude and phase adjustment information, for each channel, through which the beam directivity of the matrix to achieve control. This method based on digital beamformer set to extract selection method can effectively improve the complex environment of airspace digital matrix pointing special sound field digital matrix capabilities (such as 3D stereo field, virtual surround sound field, directivity sound field, etc) the generation effect offers a reliable way to achieve.
    The above embodiments are only used to illustrate the technical aspect of the present invention, not by way of limitation. While the embodiment of the present invention with reference to a detailed description of those skilled in the art should understand that the technical solution of the present invention, modifications and substitutions, which should fall within the claimed scope of the present invention, require.
    权利要求:
    Claims (25)
    [0001]
    1. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM characterized by comprising the steps of: (1) digital format conversion, to convert the original signals to digital signals based on PCM encoding; (2) Channel equalization processing; (3) Control speaker formation; (4) The realization of ∑-Δ multi-bit modulation; (5) thermometer code conversion, for converting low-bit encoded PCM signals with a bit width of M into unary code vectors of a digital power amplifier and a transducer load corresponding to 2M transmission channels; (6) dynamic mismatch shaping processing to reorder the thermometer encoded vectors; and (7) extracting channel information and sending it to the digital amplifier and directing sound load.
    [0002]
    2. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 1 and further characterized in that the original signals to be converted in step (1) are analog signals which in step (1) are in first converted to digital signals based on PCM encoding by analog to digital conversion, and then converted in terms of parameter requirements of a designated bit width and a sampling rate into PCM encoded signals that meet the demands of the parameters.
    [0003]
    3. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 1 and further characterized in that the original signals to be converted in step (1) are digital signals which in step (1) are converted in PCM encoded signals in terms of parameter requirements of a designated bit width and a sampling rate.
    [0004]
    4. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 1 and further characterized in that the channel equalization in step (2) is processed by an equalizer with parameters obtained by measurement and calculation.
    [0005]
    5. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 1 and further characterized in that the speaker formation in step (3) is controlled by a speaker trainer with a channel weight coefficient calculated by a normal method for speaker formation using the following formula (1):
    [0006]
    6. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 1 and further characterized in that the multi-bit modulation process ∑-Δ in step (4) is as follows: interpolation filtering of a “high-bit” PCM code interpolation filter after the equalization process according to a designated over-sampling factor, to obtain over-sampling PCM encoded signals; and then perform ∑-Δ modulation to compress the noise energy into the audio bandwidth, thus converting high-bit PCM code to low-bit PCM code.
    [0007]
    7. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 6 and further characterized in that the multi-bit modulation ∑-Δ in step (4) applies a noise shaping treatment at the output of the interpolation filter oversampling signals to compress the noise energy out of the audio band using either single stage higher order serial modulation method or parallel stage modulation method.
    [0008]
    8. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 1 and further characterized in that the code in each digit of the "unary" code vectors in step (5) is sent to the channel corresponding digital, the code in each digit having only two level states of "0" or "1" at any time the transducer load is turned off when in the "0" state and turned on when in the "1" state.
    [0009]
    9. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 1 and further characterized in that in the incompatibility shaping dynamic processing of step (6) the shaping algorithms including DWA (Averaging by weighted data), VFMS (return vector mismatch conformation) and/or TSMS (tree structure mismatch conformation) are used to shape the frequency spectrum of non-linear harmonic distortion arising from the frequency response difference between elements of the array, to reduce the amplitude of the in-band harmonic distortion components and compress the associated power to the high-frequency out-of-band section.
    [0010]
    10. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 1 and further characterized in that the channel information extraction in step (7) performs a distribution of coded information for each channel , in which the signal processing is as follows: first the dynamic mismatch formatter of each channel performs the dynamic mismatch shaping to obtain reordered shaping vectors, and then a designated code digit is selected from the 2M digits of the shaping vector of each channel as channel code according to a certain extraction selection rule, in which in order to ensure the information is completely restored the selected digit number of one channel is different from that of other channels and all numbers digits of all 2M channels completely contain the order of digits from 1 to 2M.
    [0011]
    11. CHANNEL EQUALIZATION AND BEAM CONTROL METHOD FOR A DIGITAL SPEAKER SYSTEM as claimed in 10 and further characterized in that in the channel information extraction process the digit selection is performed according to a simple selection rule. no. i of the channel and coded digit information of the same no. i.
    [0012]
    12. CHANNEL EQUALIZATION METHOD AND CONTROL OF THE !EI"E #FOR A HIGH$!DI%ITALS SYSTEM as claimed in 1 plus &'&() *)& the load to be actuated in step (7 ) may be a digital speaker arrangement that includes a plurality of speaker units, or a speaker unit that has multiple voice coil windings, or a digital speaker arrangement containing a plurality of units of multi-voice coil speaker.
    [0013]
    13. HIGH$!DI%ITALS SYSTEM WITH CHANNEL EQUALIZATION AND !EI"ES &'&() CONTROL FEATURES *)& understand: A sound source (1), which is information to be reproduced by the system; A digital converter (2), which is electrically coupled to the output end of said sound source (1), to convert input signals into high-bit PCM encoded signals with a bit width of N and a rate of fs sampling; A channel equalizer (3), which is electrically coupled to the output end of the digital converter (2), to perform an inverse filtering equalization on the frequency response of each channel to eliminate frequency response fluctuation in the band of each channel; A beam generator (4), which is electrically coupled to the output end of the channel equalizer (3), to control the spatial domain of the beam emitted shape of the speaker array and create a distribution of Characteristic sound field as a 3D stereo sound field, d field. and virtual surround sound and directional sound field and the like, to achieve the purpose of reproducing special sound effects; A ∑-Δ modulator (5), which is electrically coupled to the output end of said beam generator (4), for performing over-sampling interpolation filtering and ∑-Δ multi-bit code modulation to obtain low-bit PCM encoded signals with a reduced bit width; A thermometer encoder (6), which is electrically coupled to the output end of said ∑-Δ modulator (5), for converting the low-bit PCM encoded signals into unary code vectors that are in equal quantity to the system's digital channels , thus digitizing the channel change control vectors; A dynamic mismatch conformer (7), which is electrically coupled to the output end of said thermometer encoder (6), to eliminate non-linear harmonic distortion components of spatial domain synthetic signals arising from the frequency response difference between the elements of the arrangement, reducing the amplitude of the in-band harmonic distortion components, and compressing the power of the harmonic frequency components for the high frequency section out of the band, thus reducing the amplitude of in-band harmonic distortion and improving the quality of the sound of encoded signals ∑-Δ; an extract selector (8), which is electrically coupled to said dynamic mismatch formatter (7), to extract certain digital encoded information from the conformation vectors of each channel and control the information of the channel's on/off action; A multi-channel digital amplifier (9), which is electrically coupled to said extraction selector (8), to amplify the power of coded control signals of each channel, and trigger the on/off action of the post-stage digital load, and an arrangement and; A digital load arrangement (10), which is electrically coupled to the output end of the multi-channel digital amplifier (9), to obtain electro-acoustic conversion and conversion of digital electrical signals into air vibration signals in analog format.
    [0014]
    14. HIGH$!DI%ITALS SYSTEM WITH CHANNEL EQUALIZATION AND !EI"ES CONTROL FEATURES as claimed in 13 plus &'&() *)& the sound source (1) comprises analog signals or encoded digit signs.
    [0015]
    15. HIGH$!DI%ITALS SYSTEM WITH CHANNEL EQUALIZATION AND !EI"ES CONTROL FUNCTIONS as claimed in 13 and further characterized in that the digital converter (2) contains analog to digital converter, digital interface circuits, such as USB, LAN, COM or the like, and the interface protocol program
    [0016]
    16. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the channel equalizer (3) performs equalization processing in terms of reverse filtering response parameters in the domain of the time or frequency domain, to eliminate the frequency response fluctuation in the band of each channel and correct the difference in the frequency response of the channels.
    [0017]
    17. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the beam generator (4) performs processing for transmitted signals from each channel, using the projected weighted vectors, to regulate their amplitude and phase information.
    [0018]
    18. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the processing of the ∑-Δ modulator signals (5) is as follows: firstly the PCM signals encoded with a bitwidth of N and a sampling rate of fs are subjected to oversampling interpolation filtering according to the oversampling factor to obtain the PCM encoded signals with bitwidth of N and a sampling rate of osmf and then PCM encoded signals with a bit width of N are converted to low-bit PCM encoded signals with a bit width of M (M <N).
    [0019]
    19. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the ∑-Δ (5) modulator performs noise shaping on the output of oversampling signals from the interpolation filter to compress the out-of-band noise energy, in terms of the structure of the higher order single stage serial modular or multi-stage parallel modulator structure.
    [0020]
    20. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the thermometer encoder (6) is used to convert the low-bit PCM encoded signals with a bit width M in digital amplifier unary code signal vectors and transducer load corresponding to the 2M channels, the code information of each digit of the unary code vectors being assigned to a corresponding digital channel to bring the transducer load into the coding stream of signal and achieve digital coding and digital switch control for transducer load.
    [0021]
    21. DIGITAL SPEAKER SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the dynamic mismatch conformer (7) uses conforming algorithms including DWA (weighted average data), VFMS (conformation of return vector mismatch) and/or TSMS (tree structure mismatch conformation) to conform the frequency spectrum of nonlinear harmonic distortion arising from the frequency response difference between the array of elements, to reduce the amplitude of the components distortion harmonics in band and compress the power thereof to the high frequency band section, thus reducing the amplitude of harmonic distortion in band.
    [0022]
    22. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the extraction selector (8) extracts according to a certain rule for extracting one-digit information from the vectors of shaping each of the 2M digital channels as the coded information output of the corresponding channel, to control the on/off action of the post-phase transducer load.
    [0023]
    23. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the digital multi-channel amplifier (9) sends the output of the extraction selector (8) switching signals to one end Grid MOSFET of a full bridge power amplification circuit, thus the on/off action of the power supply circuit to the load to be controlled by the on/off state of the MOSFET.
    [0024]
    24. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FEATURES as claimed in 13 and further characterized in that the digital array load (10) is a digital array comprising a plurality of speaker units, each digital channel consists of one or more speaker units; or a multi-voice coil speaker unit, each digital channel of which consists of one or more voice-coils; or an array of multiple voice coil speakers, each digital channel of which consists of multiple voice coils and multiple speaker units.
    [0025]
    25. DIGITAL SPEAKERS SYSTEM WITH CHANNEL EQUALIZATION AND BEAM CONTROL FUNCTIONALITIES as claimed in 13 or 24 and further characterized in that the configuration of the load arrangement of the digital arrangement (10) is according to the number of transducer units and the demand for practical application.
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    同族专利:
    公开号 | 公开日
    CA2853294C|2017-09-12|
    JP6073907B2|2017-02-01|
    CN102404672B|2013-12-18|
    CA2853294A1|2013-05-02|
    EP2587836A1|2013-05-01|
    CN102404672A|2012-04-04|
    US20130108078A1|2013-05-02|
    US9167345B2|2015-10-20|
    JP2014535205A|2014-12-25|
    BR112014009896A2|2017-04-18|
    KR101665211B1|2016-10-11|
    KR20140084193A|2014-07-04|
    WO2013060077A1|2013-05-02|
    EP2587836B1|2016-03-23|
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    2017-04-25| B08F| Application fees: application dismissed [chapter 8.6 patent gazette]|
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    2019-10-08| B06U| Preliminary requirement: requests with searches performed by other patent offices: procedure suspended [chapter 6.21 patent gazette]|
    2021-05-04| B09A| Decision: intention to grant [chapter 9.1 patent gazette]|
    2021-06-22| B16A| Patent or certificate of addition of invention granted|Free format text: PRAZO DE VALIDADE: 20 (VINTE) ANOS CONTADOS A PARTIR DE 28/12/2011, OBSERVADAS AS CONDICOES LEGAIS. |
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    CN2011103311009A|CN102404672B|2011-10-27|2011-10-27|Method and device for controlling channel equalization and beam of digital loudspeaker array system|
    CN2011103311009|2011-10-27|
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