WO2013060077A1 - 数字化扬声器阵列系统的通道均衡与波束控制方法和装置 - Google Patents

数字化扬声器阵列系统的通道均衡与波束控制方法和装置 Download PDF

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Publication number
WO2013060077A1
WO2013060077A1 PCT/CN2011/084794 CN2011084794W WO2013060077A1 WO 2013060077 A1 WO2013060077 A1 WO 2013060077A1 CN 2011084794 W CN2011084794 W CN 2011084794W WO 2013060077 A1 WO2013060077 A1 WO 2013060077A1
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Prior art keywords
channel
digital
signal
array
bit
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PCT/CN2011/084794
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English (en)
French (fr)
Inventor
马登永
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苏州上声电子有限公司
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Application filed by 苏州上声电子有限公司 filed Critical 苏州上声电子有限公司
Priority to KR1020147013027A priority Critical patent/KR101665211B1/ko
Priority to JP2014537450A priority patent/JP6073907B2/ja
Priority to BR112014009896-4A priority patent/BR112014009896B1/pt
Priority to CA2853294A priority patent/CA2853294C/en
Publication of WO2013060077A1 publication Critical patent/WO2013060077A1/zh

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/005—Details of transducers, loudspeakers or microphones using digitally weighted transducing elements
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R1/00—Details of transducers, loudspeakers or microphones
    • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/403—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers loud-speakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/12—Circuits for transducers, loudspeakers or microphones for distributing signals to two or more loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
    • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
    • H04R2201/403—Linear arrays of transducers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2203/00—Details of circuits for transducers, loudspeakers or microphones covered by H04R3/00 but not provided for in any of its subgroups
    • H04R2203/12—Beamforming aspects for stereophonic sound reproduction with loudspeaker arrays
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2205/00—Details of stereophonic arrangements covered by H04R5/00 but not provided for in any of its subgroups
    • H04R2205/022—Plurality of transducers corresponding to a plurality of sound channels in each earpiece of headphones or in a single enclosure
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2430/00—Signal processing covered by H04R, not provided for in its groups
    • H04R2430/20—Processing of the output signals of the acoustic transducers of an array for obtaining a desired directivity characteristic
    • H04R2430/23—Direction finding using a sum-delay beam-former

Definitions

  • the present invention relates to a channel equalization and beam steering method and apparatus, and more particularly to a channel equalization and beam steering method and apparatus for a digital speaker array system.
  • the speaker system R&D is gradually moving towards low power consumption, small form factor, digitization and integration.
  • the digitization process of the speaker system has been advanced to the power amplifier, but in the latter stage of digital power amplifier, it is necessary to use high-quality inductors with large volume and high cost.
  • the capacitor performs passive analog low-pass filtering to remove high frequency carrier components to demodulate the original analog signal.
  • the US patent discloses a digital speaker system implementation method based on PWM modulation technology and class-BD power amplifier technology.
  • this digital speaker system based on PWM modulation technology has two disadvantages: 1
  • the coding method based on PWM modulation technology has inherent nonlinear defects due to its modulation hook, which causes the coded signal to generate nonlinearity in the desired frequency band.
  • the distortion component if further improved by linearization means, the difficulty and complexity of the modulation method will be greatly improved.
  • the PWM modulation method itself has a low frequency, which is generally in the frequency range of 200 K Hz - 400 KHz, which makes the signal-to-noise ratio of the encoded signal not limited by the sample rate. Got further improvement.
  • Chinese patent CN 101803401A discloses a multi-bit based Delta-modulated digital speaker system, through multi-bit ⁇ -
  • the ⁇ modulation and temperature if coding technology converts the high-bit PCM code into a one-ary code vector, which is used as a control vector for controlling the switching action of the speaker array, and eliminates the spatial synthesis introduced by the difference of the frequency response of the array element through the dynamic mismatch shaping technique.
  • the higher harmonic component of the signal although the patent realizes the full digitization of the entire signal transmission link of the system, and relies on the dynamic mismatch shaping technique, the total harmonic distortion ratio of the spatial composite signal is reduced, but this dynamic mismatch
  • the shaping technique does not balance the frequency fluctuations in the channel audio band. Therefore, the internal frequency response fluctuation of each channel of the audio channel causes a large deviation between the spectrum of the system restoration signal and the real spectrum of the source signal, which results in a restored sound field.
  • the beam steering method for the base channel extension adjustment disclosed in the Chinese patent CN 101803401A only adjusts the phase information of the transmission signals of each channel of the array, and does not consider the amplitude adjustment of the transmission signals of each channel, and belongs to a relatively simple beamforming.
  • the method has weak beam steering capability and has certain beam steering capability only in an environment close to a free field.
  • the delay is based on the delay.
  • the method of control will not be able to complete the steering control of multiple beams.
  • Only relying on the channel-based control method can not achieve better beam pointing control. Therefore, for the beam directivity control problem of the T-speaker array in the reverberant environment, it is necessary to find a complex beamforming method with anti-reverberation capability.
  • the transmission signal of the channel is simultaneously adjusted in amplitude and phase to achieve the desired sound field control effect.
  • the digital speaker array system requires application in frequency band flatness and beam pointing, and a digital speaker array system device with channel equalization function and beam steering function is fabricated.
  • the object of the present invention is to overcome the shortcomings of the existing digital system in channel equalization, and propose a channel equalization and beam control method for a digital speaker array system and a digital speaker system device with channel equalization and beam control functions.
  • an aspect of the present invention provides a channel equalization and beam control method for a digital speaker array system, including the following steps:
  • thermometer code conversion converting a low bit PCM encoded signal having a bit width M into a unicode vector corresponding to a digital power amplifier and a transducer load of 2 M channels;
  • the digital format conversion described in the step 1) is divided into two cases of analog and digital signals.
  • the analog signal the first need to undergo analog-to-digital conversion, convert to a digital signal based on PC encoding, and then according to the specified bit.
  • the parameters of the width and sampling rate are required to be transformed and converted into PCM coded signals satisfying the parameter requirements.
  • the parameters of the specified bit width and sampling rate need to be transformed and converted into PCM coded signals satisfying the parameter requirements.
  • the parameters of the equalizer can be obtained according to the measurement method. Assuming that the number of measurement elements is the number of measurement points in the desired position area is M, the array element emits a white noise signal by acquiring the received signal r(0 at the measurement point, and calculating the impulse mh j of the array element channel to the desired measurement position point, where For the index number of the i-th array element, .
  • the fitting may be obtained using weighted average of the first array element into a desired region of the impulse response W wherein; weight vector is the frequency response of the array element to the second measuring points; Nilv algorithm then estimated wave calculated The inverse filter response of the average impulse response ⁇ — ⁇ ; finally selects the average of the first array element to the desired position area Then, by setting the compensation factor h s , the inverse filter response of the other remaining array elements - 1 (2 ⁇ ⁇ N), after compensation, the compensation result 1 It is exactly the same as the reference vector, so that the response vector of the equalizer is obtained.
  • the channel weight coefficient of the beamformer can be used to calculate the channel weight coefficient according to the conventional beamforming design method. Assuming that the number of array elements in the array is N, the steering vector of the spatial domain is: a The desired spatial beam shape is: ⁇ , ⁇ .
  • the amplitude and phase adjustments are performed on the transmission signals of the respective channels by using the array weight vector, thereby guiding the spatial acoustic radiation beams of the array into a desired region.
  • the multi-bit ⁇ modulation described in step 4) is processed as follows: First, the equalization processed high-bit PCM coding is subjected to interpolation filtering processing according to a specified oversampling factor by an interpolation filter to obtain oversampling. The PCM coded signal; then, ⁇ - ⁇ modulation process is performed to push the noise energy in the audio bandwidth range beyond the audio band, ensuring that the system has a sufficiently high signal-to-noise ratio in the audio band, while After the ⁇ modulation process, the original high-bit PCM code is converted into a low-bit PCM code, and the number of bits of the PCM code is reduced.
  • the multi-bit sigma-delta modulation in step 4) adopts a ⁇ - ⁇ modulation method according to various existing ⁇ - ⁇ modulation methods - like a high-order single-stage (Higher ⁇ Order Single- Stage) serial modulation method or multi-stage (Cascade, MASH) parallel modulation method, which performs noise shaping processing on the oversampled signal outputted by the interpolation filter, and pushes the noise energy out of the audio band, ensuring The system has a high enough in-band signal to noise ratio.
  • thermometer code conversion described in step 5 is for converting a low bit PCM coded signal having a bit width M into a unicode vector corresponding to the digital power amplifier and the transducer load of the channels.
  • the code on each digit of the unary code vector is sent to the corresponding digital channel, and the encoding on each digit has only two levels of "0" and "1" at any time, in the "0" state.
  • the transducer load is turned off, the transducer load is turned on in the "1" state.
  • the temperature ⁇ encoding operation is used to distribute the encoded information to multiple transducer load channels, and the transducer load is incorporated into the signal encoding process to implement digital encoding and digital switching control of the transducer array.
  • the dynamic mismatch shaping processing in step 6) is used to reorder the temperature chirp coding vector, further optimize the data distribution scheme of the unary code vector, and eliminate the spatial synthesis signal caused by the frequency difference between the Chen and the Yuan. Nonlinear harmonic distortion component.
  • the dynamic mismatch shaping processing in step 6) adopts various existing shaping algorithms such as DWA (Data-Weighted Averaging) and VFMS (Vector-Feedback mismatch-shaping vector). Feedback mismatch shaping) and TSMS (Tree-Structure mismatch-shaping
  • DWA Data-Weighted Averaging
  • VFMS Vector-Feedback mismatch-shaping vector
  • TSMS Trae-Structure mismatch-shaping
  • the shaping operation reduces the intensity of the in-band harmonic distortion component and pushes its power to the out-of-band high frequency band, which reduces the in-band harmonic distortion intensity and improves the sound quality level of the ⁇ - ⁇ coded signal.
  • the channel information is extracted in step 7), and the coding information distribution operation is performed on each channel.
  • the signal processing process is as follows: As shown in FIG. 4, first, the dynamic mismatcher of each channel performs dynamic mismatch shaping processing. After shaping, the shaping vector of the bit sequence update is obtained; then, according to a specific sleeve selection criterion, a specified digit encoding is selected from the 2 M digits of the shaping vector of each channel as the output encoding of the channel. In order to ensure the complete restoration of information, the digits selected by each channel cannot be repeated, and all the digits selected by 2 M channels completely contain 1 to 2 M digits.
  • the digital selection operation can generally be performed according to the simple criterion of the i-th channel from which the i-th digital coded information is selected from the shaping vector.
  • the preset equalization and beam weighting processing on multiple array channels are effectively inherited, thus providing equalization and directivity control operations for digital arrays. An effective way to achieve it.
  • the load in step 7) may be a digital speaker array composed of a plurality of speaker units, a speaker unit having a plurality of voice coil windings, and a digital speaker array composed of a plurality of multi-speaker speaker units.
  • Another aspect of the present invention provides a digital speaker array system device having a channel equalization and beam steering function, including - a sound source, which is information to be played by the system;
  • a digitizer coupled to the output of the sound source for converting the input signal into a high bit PCM encoded signal having a bit width N and a sampling rate;
  • a channel equalizer is connected to the output end of the digitizer to inversely filter and correct the frequency response of each channel to eliminate the in-band fluctuation of the channel frequency response;
  • a beamformer is connected to the output of the channel equalizer, and ffl controls the spatial radiation shape of the speaker array beam to generate sound field distribution characteristics such as a 3D stereo field, a virtual surround sound field, and a directional sound field, thereby achieving special sound effect playback.
  • a delta modulator coupled to the output of the beamformer for performing oversampling interpolation filtering and multi-bit ⁇ ⁇ encoding puncturing processing to obtain a bit-width reduced low-bit PCM encoded signal
  • thermometer encoder coupled to the output of the ⁇ - ⁇ modulator to convert the low-bit PC encoded signal into a unicode vector equal to the number of digital channels of the system for use in digitizing the control vector of the channel switch; a shaping device, connected to the output of the temperature encoder, for eliminating the nonlinear harmonic distortion component of the spatial synthesis signal introduced by the difference in frequency response between the elements, and the intensity of the harmonic distortion component in the bass band, Pushing the power of these harmonic components into the out-of-band high frequency band reduces the intensity of harmonic distortion in the band. Improve the sound quality level of the ⁇ coded signal;
  • a decimation selector is connected to the output end of the dynamic mismatch shaper for extracting specific digit encoding information from the shaping vector of each channel for controlling the channel control information for the opening/closing action of the channel;
  • a multi-channel digital power amplifier is connected to the output of the armature selector, and is configured to power amplify the control coded signals of each channel for driving the digital load of the subsequent stage to be turned on/off;
  • a digital array load is coupled to the output of the multi-channel digital power amplifier for performing an electroacoustic conversion operation to convert the digitized electrical signal to an air vibration signal in an analog format.
  • the sound source may be an analog signal or a digitally encoded signal, and may be derived from an analog source signal generated by various analog devices, or may be a digitally encoded signal generated by various digital devices.
  • the digitizer can include digital interface circuits and interface protocol programs such as analog-to-digital converters, USB, LAN, COM, etc., and can be compatible with existing digital interface formats. Through these interface circuits and protocol programs, digital speakers The array system device can flexibly and conveniently exchange and transmit information with other device devices.
  • the original input analog or digital sound source signal is converted into a high bit with a bit width of N and a sampling rate. PCM encoded signal.
  • the channel equalizer can perform equalization operation according to the inverse filtering response parameter in the time domain or the frequency domain, and eliminate frequency fluctuation fluctuations in the audio band of each channel; and ⁇ , also corrects the frequency response difference of each channel, so that each channel The frequency response is consistent.
  • the beamformer performs weighting processing on the transmission signals of each channel by using the weight vector provided, and adjusts the amplitude and phase information thereof, so that the spatial pattern of the digitized array in a complex environment reaches the desired requirement.
  • the ⁇ - ⁇ modulator has the following signal processing: First, the PCM code with the original bit width N and the sampling rate of . /; is oversampled. Perform interpolation filtering processing to obtain a bit width of N and a sampling rate? The PCM coded signal is then converted into a low bit PCM coded signal having a bit width of M (M ⁇ N) according to a multi-bit ⁇ ⁇ modulation mode, thereby reducing the PCM coded signal. Bit width.
  • the ⁇ modulator can perform oversampling signals of the interpolation filter output according to a signal processing structure of various existing ⁇ modulators such as a high-order single-stage serial modulator structure or a multi-stage parallel modulator structure.
  • the noise shaping process pushes the noise energy out of the audio band, ensuring that the system has a sufficiently high in-band signal-to-noise ratio.
  • thermometer encoder converts the low-bit PC coded signal having a bit width of h4 into a one-ary code signal vector corresponding to a digital power amplifier and a transducer load of 2 M channels, and each digital coded information of the unary code vector is distributed. To a corresponding digital channel, thereby incorporating the transducer load into the signal encoding process, Digital encoding and digital switching control of the transducer load is achieved.
  • the dynamic mismatch shaper performs a shaping operation on the nonlinear harmonic distortion spectrum introduced by the frequency difference between the array elements by using various existing shaping algorithms such as DWA, VTMS and TSMS algorithms, and lowers the in-band
  • the intensity of the harmonic distortion component pushes its power to the out-of-band high frequency band, thereby reducing the intensity of the harmonic distortion in the band and improving the sound quality level of the ⁇ - ⁇ coded signal.
  • the decimator selects each channel of the two digital channels into a vector according to a specific decimation criterion, and the sleeve takes a digital information as the output coding information of the channel for controlling the transducer load of the subsequent stage. Turn on/off. After the bit-sleeve taking and merging operation of the armature selector, the original equalizer response and channel directivity weighting vector operation of multiple channels are effectively realized, which ensures the frequency response flatness and beam direction of the digital array. Controllability.
  • the multi-channel digital power amplifier sends the switching signal of the decimator output to the gate end of the MOSFET of the full-bridge power amplifier circuit, and controls the turn-on and turn-off of the MOSFET to control the turn-on and turn-off of the power supply to the load power supply. , thereby achieving power amplification of the digital load.
  • the digital array load may be a digital array composed of a plurality of speaker units, a speaker unit of a plurality of voice coils, or a display composed of multi-voice speakers.
  • Each digital channel of the digitized load may consist of a single or multiple speaker units; it may also consist of a single or multiple voice coils; it may also be composed of multiple voice coils and multiple speaker units.
  • the shape of the array of digitized loads can be arranged according to the number of transducer units and the actual requirements of the 3 ⁇ 4, to form various array shapes suitable for the actual demand.
  • the invention has the advantages that: the whole signal transmission link of the system is fully digitized, and the whole system device is completely composed of digital devices, which facilitates the high-level integrated circuit design and improves the system work. Stability, reducing the power consumption, volume and weight of the system; At the same time, the digital speaker array system can flexibly and conveniently interact with other digital system equipment to better adapt to the development requirements of digitalization.
  • the multi-bit ⁇ ⁇ modulation technology pushes the noise power in the audio band to the out-of-band high-frequency region through the noise shaping method, and ensures the high signal-to-noise ratio requirement in the audio band.
  • the hardware implementation circuit of the modulation technology is simple. It has good immunity to parameter deviations generated during the fabrication of circuit devices. In addition,
  • the implementation of the full number of sub-systems adopted by the invention has stronger anti-interference ability and can ensure stable and reliable work in a complex electromagnetic interference environment.
  • the dynamic mismatch shaping algorithm used in the present invention can effectively reduce the nonlinear harmonic distortion intensity introduced by the frequency difference between the array elements and improve the sound quality level of the system, so the system is for the transducer unit.
  • the frequency response deviation between the two has good immunity.
  • each transducer unit is assigned a corresponding unary code signal, so that each speaker The unit (or each voice coil) works in the on or off state, and the state of the alternate switch operation effectively avoids the overload distortion of each speaker unit (or each voice coil), thereby extending the speaker units (or The service life of each voice coil; at the same time, the transducer works in a switch mode, the electroacoustic conversion efficiency is higher, and the transducer has less heat.
  • I Digital power amplifier circuit directly sends the amplified switch signal to the speaker terminal to control the speaker to perform the on and off operations. It is not necessary to add a large volume and expensive inductor and capacitor to the analog power supply in the digital amplifier stage. , reducing the size and cost of the system; at the same time, for the piezoelectric transducer load with capacitive characteristics, it is usually necessary to add an inductor for impedance matching to increase the output sound power of the piezoelectric speaker, and apply it at the transducer end.
  • the digital signal, its impedance matching effect is better than the traditional impedance matching effect of applying analog signal on the transducer end.
  • thermometer coding method is such that the unary code signal allocated by each group of array elements only contains part of the information components of the original source signal, and the information radiated by the single array element alone cannot complete the complete restoration of the sound source information, and only all the groups are combined.
  • the synthesis of the radiated sound field in the space of the array element can completely restore the information of the sound source; the synthesis of the spatial sound field of the combined multi-array element completes the information restoration mode, and the restored information has spatial directionality on the axis of symmetry of the array. With the largest signal to noise ratio, the farther away from the axis, the lower the signal to noise ratio.
  • the channel equalization method can keep the internal frequency response of each channel audio level flat, and correct the frequency response difference between the channels, ensuring that the spectrum of the system restored sound source signal and the original spectrum of the original source signal are consistent, thus ensuring the digital weight.
  • the system actually reproduces the sound field effect of the original sound source; at the same time, the equalization method brings the internal frequency response flatness of each channel and the consistency of the frequency response between the channels, which has better stability for various adaptive algorithms. Faster convergence speed and better robustness provide favorable support.
  • Channel equalization method based on data extraction selection can better suppress the frequency response fluctuation of each channel, improve the sound field reduction quality of the digital system, and eliminate the large frequency response difference between channels, so After the channel equalization processing, the frequency response deviation between the channels is compensated to a large extent, leaving only a small amount of residual deviation. These residual deviations can further rely on the mismatch shaping algorithm for better correction processing, thus mismatch shaping. The ability of the algorithm to remove small deviations is also effective. After the channel equalization processing, the frequency response difference of the array elements is well corrected, thus ensuring that various display beam control algorithms based on coherent accumulation of array element channels can be effectively operated.
  • the digital array beamforming method based on data extraction selection can effectively improve the spatial sound field control capability of the digital array in a complex environment.
  • the beam control method ensures that the digital speaker array has better beam directivity in a complex environment, and the conventional beam steering method can be well applied to the beam of the digital array through the information combination method of the sleeve selection.
  • Control provides an effective way to generate effects for special sound fields (such as 3D stereo fields, virtual surround sound fields, directional sound fields, etc.) in real environments.
  • K. Data extraction selection method which can directly extend the traditional channel equalization and beamforming algorithms based on PCM coding format to digital array systems based on multi-bit ⁇ - ⁇ modulation, thus being used for traditional channel equalization and beam control algorithms.
  • a bridge between digital array systems based on multi-bit ⁇ - ⁇ spurs ensures that traditional algorithms can continue to perform effective channel equalization and beam steering in array systems based on multi-bit ⁇ - ⁇ modulation.
  • FIG. 1 is a block diagram showing the components of a digital speaker system having channel equalization and beam steering functions in accordance with the present invention
  • FIG. 2 is a schematic diagram showing channel parameter measurement in the channel equalization parameter estimation process of the present invention.
  • FIG. 3 is a schematic diagram showing channel weight vector loading in the beam steering process of the present invention.
  • FIG. 4 is a schematic diagram showing the extraction rule of the present invention in the channel information extraction process
  • Figure 5 is a graph showing the amplitude spectrum of the inverse filter used in the channel equalization process according to an embodiment of the present invention
  • Figure 6 is a diagram showing the signal processing of the fifth-order CiFB modulation structure employed by the ⁇ modulator according to an embodiment of the present invention.
  • Figure ⁇ shows a schematic diagram of the opening control of the thermometer code vector of an implementation of the present invention
  • FIG. 8 is a flow chart showing the signal processing of the VFMS mismatch shaping algorithm used by the dynamic mismatch shaper according to an embodiment of the present invention.
  • Figure 9 is a diagram showing the extraction criteria used by the armature selector of an embodiment of the present invention.
  • Figure 10 is a schematic view showing the arrangement of an 8-element speaker array according to an embodiment of the present invention.
  • FIG. 11 is a schematic view showing the arrangement position of the speaker array and the microphone unit of an embodiment of the present invention
  • FIG. 12 is a view showing the comparison of the amplitude spectrum curves of the system frequency response of the position point of the array axis at 1 meter before and after the equalization according to an embodiment of the present invention; ;
  • Figure 13 is a view showing a beam pattern curve generated in three predetermined directions of -60 degrees, 0 degrees, and +30 degrees in an embodiment of the present invention
  • Fig. 14 shows the parameter values employed by the ⁇ modulator according to an embodiment of the present invention.
  • the present invention first converts the sound source signal in the audible sound to a high bit PCM encoded signal having a bit width of N by digital conversion; Perform inverse filtering equalization processing on the digital sound source signals of each channel, eliminate frequency fluctuations in the audio band of each channel, and eliminate the difference in frequency response between channels; then use beamforming technology to weight the equalized signals of each channel.
  • the subharmonic component reduces the total harmonic distortion of the system and improves the sound quality level of the system.
  • the bit information on one digit is extracted from the mismatch shaping vector of each channel, and the number sent to the channel is sent to the channel.
  • the power amplifier forms a power signal, and drives the digital load of the channel to be turned on or off.
  • the spatial sound field radiated by the digital load of all channels is superimposed to restore the source signal in a predetermined area of the space.
  • a digital speaker system device with channel equalization and beam steering function As shown in FIG. 1, a digital speaker system device with channel equalization and beam steering function according to the present invention is constructed, and the main body thereof is composed of a sound source 1, a digitizer 2, a channel equalizer 3, a beam former 4, and a ⁇ modulation.
  • the device 5, the temperature encoder 6, the dynamic mismatch shaper 7, the decimation selector 8, the multi-channel digital power amplifier 9 and the digital array load 10 are composed.
  • Sound source 1 you can use the MP3 format audio file stored in the hard disk of the PC, which can be output in digital format through the USB port: You can also use the audio source file stored in the MP3 player to output in analog format; you can also use the signal source to generate Test signals within the audio range are also output in analog format.
  • the digitizer 2 is connected to the output end of the sound source 1, and includes two input interfaces of a digital input format and an analog input format.
  • a USB interface chip of the PCM2706 model of the Ti company is used, and the PC can be used.
  • the MP3 type file stored in the machine is read into the FPGA core of the model Cyclone III EP3C80F484C8 through the USB port according to the 6-bit width and 44.1 KHz sampling rate through the I2S interface protocol.
  • Analog Devices is used for the analog input format.
  • An analog-to-digital conversion chip of the AD1877 converts the analog source signal into an i6-bit, 44.1 KHz PCM encoded signal, which is also read into the FPGA chip in real time through the I2S interface protocol.
  • the channel equalizer 3 is connected to the output end of the digitizer 2, and the inverse filter parameters of each channel are calculated according to the measurement manner.
  • FIG. 5 shows the inverse filter amplitude spectrum curves of the channels 1 to 8, according to The inverse filter parameters are used to equalize each channel to obtain an equalized 16-bit, 44. 1 KHz sampling rate PCM ⁇ ⁇ ! J.
  • a beamformer 4 connected to the output of the channel equalizer 3, calculating a weight vector of the 8-element array according to a desired beam pattern, and then loading the calculated weight vector into the FPGA through the multiplier unit
  • a ⁇ - ⁇ modulator 5 is connected to the output of the beamformer 4, first, in the FPGA chip, an oversampling interpolation filtering operation is performed, and 44.: ⁇ ⁇ , 16-bit PCM encoded signal is pressed Third level The upsampling interpolation process is performed.
  • the first-order interpolation factor is 4, the sampling rate is increased to ⁇ 6,4 ⁇ , the second-order interpolation factor is 4, the sampling rate is increased to 705.6 ⁇ , the third-order interpolation factor is 2, and the sampling rate is increased to 1411.2 ⁇ ⁇ .
  • the original 44.1 ⁇ , 16-bit PCM signal is converted to 1,41] 2 MHz, ⁇ 6-bit oversampled PCM signal; then according to the 3-bit ⁇ - ⁇ modulation method,
  • the 1.4112 MHz, 16-bit PCIV [coded signal is converted into a 1.4112 MHz > 3-bit PCMb coded signal.
  • the ⁇ - ⁇ modulator uses a 5th-order CIFB Cascaded Integrators.
  • the topology of the modulator is shown in Table 1.
  • the shift addition operation is usually used instead of the constant multiplication operation, and
  • the parameters used by the - ⁇ modulator are represented by CSD code.
  • the temperature encoder 6 is connected to the output of the ⁇ modulator 5, and converts the 1,4112 MHz, 3-bit sigma-delta modulated signal into a 1.4112 MHz, one-bit code with a bit width of 8 according to a thermometer encoding method. . As shown in Figure 7, when the 3-bit PCM is encoded as "001", the converted thermometer is encoded as "00000001.
  • This code is used to control the opening of one of the arrays of the transducer array, and the remaining 7 elements are turned off;
  • the converted thermometer code is "00001111”, which is used to control the opening of the 4 elements of the transducer display, and the remaining 4 elements are turned off;
  • the 3-bit PCM code is "111” ", the converted thermometer is coded as "01111111”, which is used to control the 7 array elements of the transducer array to be turned on, leaving only 1 array element off.
  • the dynamic mismatch shaper 7 is connected to the output of the thermometer encoder 6 for eliminating nonlinear harmonic distortion components caused by the difference in frequency response between the elements.
  • the dynamic mismatch shaper 7 sorts the 8-bit temperature if code according to the optimization criterion of the minimum nonlinear harmonic distortion component, thereby determining the coding allocation mode for the 8 transducer elements, as shown in FIG.
  • the code distribution method performs the switching control of the transducer array, and will contain the least harmonic distortion component in the signal synthesized by the display radiation sound field.
  • the dynamic mismatch shaper adopts the VFMS algorithm, and its signal processing flow is as shown in FIG. 8.
  • the thick line represents a V-dimensional vector
  • the thin line represents a scalar
  • the input signal V is a ⁇ - ⁇ modulator
  • the W-dimensional code vector processed by the thermometer encoder the code vector includes V "]" states and ⁇ "" ⁇ "0" states
  • the output signal SV is an N-dimensional column vector after the mismatch shaping process.
  • mismatch shaping process the order of the "1" state and the "0" state of the output vector in the vector is adjusted, but the number of "1" states and "0” states remains unchanged, and each in the vector
  • the elements control the corresponding one of the array elements in the array to perform the on/off operation according to its state.
  • the unit selection module guarantees that the error introduced by the difference in frequency response can be spectrally enabled by a selection strategy A better shaping effect is obtained.
  • the -min() module indicates that the element with the smallest value is selected in the N-dimensional vector and is negative at the same time.
  • the scalar element obtained by the operation of the -minG module is u, and mtf is a mismatch shaping function.
  • the general form is 1 and M is an order.
  • the decimation selector 8 which is connected to the output of the dynamic mismatch shaper 7, is used for digital extraction in the shaping vector of each channel, and is sent to the post-stage power amplifier and the digital load. As shown in FIG. 9, each channel is subjected to a mismatch shaping process to generate an 8-ary unary code vector, and the decimation selector 7 extracts the first digit of the shaping vector according to the i-th channel, and extracts for each channel. A corresponding one-digit code signal is used as an input signal of the subsequent digital power amplifier.
  • the digital power amplifier chip selects a digital power amplifier chip of the model TAS512 from the Ti company, and the response time of the chip is on the order of 100 ns , and can transmit a one-dimensional stream signal with a distortion of 1.4112 MHz without distortion.
  • the differential input format is adopted.
  • the output data sent by the dynamic mismatch shaping is directly outputted, and the other phase is outputted by the reverse phase to form two differential signals, which are sent to the difference of the TAS5121 chip.
  • Input At the output of the amplifier, the differential output format is also applied, and the two differential signals are directly applied to the positive and negative leads of the single transducer array channel.
  • the digital display load "0" is connected to the output of the multi-channel digital power amplifier 9.
  • the digital load unit adopts a full-band speaker unit of the model B2S produced by the company, the frequency range of the unit is 270 Hz ⁇ 20 KHz, and the sensitivity (2,83V/lni) is 79 (IB, the maximum power is 2 W, rated impedance is 8 ohms M.
  • the digitized load is an 8-element speaker array, and the display is arranged in a linear array by 8 of the above-mentioned speaker units, with an array element spacing of 4 cm, and each speaker unit corresponds to one digitized channel.
  • the amplitude spectrum of the system frequency response decreases from 65 dB to 45 dB, and there is a difference of 20 dB; after applying the equalizer
  • the amplitude spectrum of the system frequency response is maintained at around 57 dB in the frequency range of 2 KHz to 20 KHz, exhibiting very flat spectral characteristics, thus ensuring the true reproduction of the synthesized signal of the system.
  • the multi-channel bit information synthesis method selected by the decimation can effectively inherit the equalizer response information of each channel, and the frequency response flatness of each channel is ensured.
  • the digital speaker array system based on channel equalization can effectively eliminate the frequency fluctuations in the audio frequency band of each channel and correct the frequency response difference between the channels, ensuring that the system has a very flat time-domain frequency response in the desired spatial region.
  • the feature ensures that the spectrum of the spatially synthesized signal of all channels can restore the true spectrum of the original source signal, ensuring that the digital playback system truly reproduces the sound field effect of the original source.
  • various adaptive spatial domain beamforming algorithms have faster convergence speed and better robustness.
  • the simulation of the array beam control is performed according to the arrangement of the speaker arrays shown in Figure il, according to the three predetermined beam main lobe directions of -60 degrees, 0 degrees and +30 degrees.
  • the array lobes are all 20 degrees wide.
  • Figure 13 shows the airspace pattern of the T array for three predetermined directions. Observing these curves, it can be seen that the beam main lobe of the array is pointing in a predetermined direction, the beam width reaches the desired requirement, and the main sidelobe amplitude difference is reached.
  • the multi-channel information synthesis method selected by the decimation can effectively inherit the amplitude and phase adjustment information of the beamformer loaded on each channel, and realize the beam directivity of the array. control.
  • the digital array beamforming method based on the extraction selection method can effectively improve the spatial directivity capability of the digital array in a complex environment, and is a digital array special sound field (such as a 3D stereo field, a virtual surround sound field, a directional sound field, etc.). Generation provides a reliable implementation path.

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Abstract

本发明公开了一种数字化扬声器阵列系统的通道均衡与波束控制方法和装置。方法包括步骤：1）数字格式转换；2）通道均衡处理；3）波束形成控制；4）多比特∑-Δ调制；5）温度计编码转换；6）动态失配整形处理；7）通道信息抽取，送至数字功放驱动阵列发声。装置包括：一音源、一数字转换器、一通道均衡器、一波束形成器、一∑-Δ调制器、一温度计编码器、一动态失配整形器、一抽取选择器、一多通道数字功放、一扬声器阵列，各单元依次连接。本发明实现了系统的全数字化，缩减了其体积、功耗、成本，提高了其电声转换效率和抗干扰能力，改善了系统可听声频带内频响平坦程度，实现了数字阵列的波束指向控制，为特殊声效的生成提供了有效的实现途径。

Description

数字化扬声器阵列系统的通道均衡与波束控制方法和装置 技术领域
本发明涉及一种通道均衡与波束控制的方法和装置， 特别涉及一种数字化扬声 器阵列系统的通道均衡与波束控制方法和装置。
背景技术
随着大规模集成电路和数字化技术的蓬勃发展，传统的模拟扬声器系统在功耗、 体积、 重量和信号传输、 存储、 处理等方面的固有缺陷越来越明显， 为了克服这些 缺陷， 扬声器系统的研发逐渐向低功耗、 小外形、 数字化与集成化的方向发展。 随 着基于 PWM调制技术的 c ass- AD型数字功放的出现， 扬声器系统的数字化进程已 经推进到功放环节， 但是在数字功放后级扔然需要借助体积较大、 成本较贵的高质 量电感和电容进行无源的模拟低通滤波操作来消除高频载波分量， 以便解调出原模 拟信号。 为了缩减数字功放的体积和成本， 实现更高程度的集成化， 美国专利 （专 利号为 US 20060049889A1、 US 20090161880AI ) 公开了基于 PWM 调制技术和 class-BD功放技术的数字化扬声器系统的实现方法。 但是这种基于 PWM调制技术 的数字化扬声器系统有两个缺点： ① 基于 PWM调制技术的编码方式， 因其调制结 钩本身具有固有的非线性缺陷， 这会造成编码信号在期望频带内产生非线性失真分 量， 如果进一步采用线性化手段进行改善的话， 其调制方式的实现难度和复杂度将 会大幅度提高。 ② 鉴于硬件实现难度， PWM调制方式本身的过釆样频率较低， 一 般在 200 K Hz - 400 KHz的频率范圈内， 这会使得编码信号的信噪比因受过釆样率 的限制而不能得到进一步提升。
针对 PWM调制技术在数字化扬声器系统实现方面存在的非线性失真和过采样 速率较低的缺陷， 并结合系统整个信号传输链路的全数字化要求， 中国专利 CN 101803401A公开了一种基于多比特 Σ - Δ调制的数字化扬声器系统， 通过多比特∑-
Δ调制和温度 if编码技术将高比特的 PCM码转换为一元码矢量， 作为控制扬声器 阵列开关动作的控制矢量， 并通过动态失配整形技术， 消除了由阵元频响差异所引 入的空域合成信号中的高次谐波分量； 该专利虽然实现了系统整个信号传输链路的 全数字化， 并依靠动态失配整形技术， 减少了空域合成信号的总谐波失真比， 但是 这种动态失配整形技术对通道音频带内的频响起伏并没有均衡作用， 因此， 这种各 通道音频带内频响起伏会引起系统还原信号频谱与音源信号真实频谱存在较大偏 差， ^而造成还原声场与真实声场的较大差异， 使得数字重放系统不能真实的再现 原来音源的真实声场效果。 另外， 这种各通道的音频带内频响起伏也会引起各种自 适应的阵列波束形成算法的稳定性变差， 收敛速度变慢， 造成自适应阵列波束形成 算法的鲁棒性变差。 目前中国专利 CN 101803401A所公幵的基干通道延 ^调整的波束导向方法，仅 调整了阵列各通道传输信号的相位信息， 并没有考虑各通道传输信号的幅度调整， 属于一种较为简单的波束形成方法， 其所产生的波束控制能力较弱， 仅在接近自由 场的环境中具有一定的波束导向能力， 在某些应用场合， 当需要数字系统产生多个 指向性波束时， 这种基于延时控制的方法将无法完成多个波束的导向控制。 另外， 在实际应用环境中， 经常会存在较多的散射边界， 使得传播信号除了直达声之外还 包含有较为丰富的多径散射信号， 在这种多径散射较为明显的混响环境中， 仅依赖 通道延 ^控制的导向方法， 无法取得较好的波束指向控制， 因此针对混响环境 T数 字扬声器阵列的波束指向性控制问题， 需要寻找具有抗混响能力的复杂波束形成方 法，对各通道的传输信号同时进行幅度和相位调整，从而达到期望的声场控制效果。
目前， 基于多比特 Σ - Δ调制的数字化阵列系统， 都是依赖于失配整形技术消除 多通道之间的频响差异性， 但是这种通道频响差异校正方法， 仅适用于少量频响偏 差的情况， 并 ϋ对相位偏差的校正能力非常弱； 另外， 失配整形技术对各通道自身 的音频带内频响起伏并不能起到均衡作 ffl , 而这些通道本身的频响起伏性会带来还 原声场的音色成份发生改变， 难于保证声场的完整恢复。 传统的数字化扬声器阵列 所采用波束控制方法是较为简单的通道延时控制方法， 这种方法仅适用于理想的自 由声场环境， 而当声场因反射或散射效应出现较为丰富的多径千扰^， 这种方法将 不再适] ¾。 在某些应用场合， 当需要阵列产生多个指向性波束， 这种基干延时控制 的方法将无法获得多个波束的声场控制效果。
针对现有的基于多比特∑- A调制的数字化扬声器阵列系统在通道均衡和波束 控制方面存在的缺陷， 需要寻找更为有效的通道均衡和波束控制方法， 以满足基于 多比特 Σ - Δ调制的数字化扬声器阵列系统在频带平坦和波束指向方面的应用需求， 并制作具有通道均衡功能和波束控制功能的数字化扬声器阵列系统装置。
发明内容
本发明的目的是克服现有数字化系统在通道均衡方面的不足， 提出了一种数字 化扬声器阵列系统的通道均衡与波束控制方法以及具有通道均衡和波束控制功能的 数字化扬声器系统装置。
为了达到上述目的， 本发明一方面提供一种数字化扬声器阵列系统的通道均衡与波 束控制方法， 包括如下歩骤：
i ) 数字格式转换,将信号转换为基于 PCM编码的数字信号；
2 ) 通道均衡处理；
3) 控制波束形成；
4) 多比特∑ Δ调制； 5)温度计编码转换，将位宽为 M的低比特 PCM编码信号转换为对应于 2M个通 道的数字功放和换能器负载的一元码矢量；
6) 动态失配整形处理， 对温度计编码矢量进行重新排序；
7) 抽取通道信息， 送至数字功放驱动负载发声。
进一步地， 歩骤 1) 中所述数字格式转换分为模拟和数字信号两种情况， 针对 模拟信号情况， 首先需要经过模数转换操诈， 转换为基于 PC 编码的数字信号， 然 后按照指定位宽和采样率的参数要求进行变换， 转换为满足参数要求的 PCM编码信 号； 针对数字信号情况， 仅需要按照指定位宽和采样率的参数要求进行变换， 转换 为满足参数要求的 PCM编码信号。
进一步地， 步骤 2) 中所述通道均衡处理， 其均衡器的参数可以按照测量方法 获取。 假定阵元数量为 期望位置区域的测量点数量为 M, 阵元发射白噪声信号 通过在测量点获取接收信号 r(0, 计算出阵元通道到期望测量位置点的冲激响 mh j ,其中 为对第 i个阵元的索引号,. /为期望区域内第 J个测量点位置的索引号; 假定第 i个阵元到) 点的冲激响应 h L/≤ii都己经计算出来， 可以根据加权 拟合的方法获得第 个阵元到期望区域的平均冲激响应 其中 w;为第 个阵元到第 个测量点的频响加权矢量； 然后再根据逆旅波器的估计算法， 计算出 平均冲激响应 的逆滤波器响应^— ^； 最后选取第一个阵元到期望位置区域的平均
Figure imgf000005_0001
, 那么通过 设置补偿因子 hs， 使其他剩余阵元通道的逆滤波器响应 —1 (2≤ ≤N) 在经过补 偿后， 其补偿结果 1
Figure imgf000005_0002
与参 考矢量 完全相同， 从而获得均衡器的响应矢量为
Figure imgf000005_0003
进一步地， 步骤 3) 中所述波束形成控制， 其波束形成器的通道权系数可以按 照常规波束形成的设计方法进行通道权系数的计算。 假定阵列的阵元数量为 N, 其 空域的导向矢量为： a 期望的空域波束形状为: θ、 <θ<θ.
Figure imgf000005_0004
假定待求的阵列权系数矢量为: W ^2 w ', 按照最小二乘准则， 可以 获得阵列权系数的^算公式如下 w = arg mm |w7 a(6?) -- D(6?1 άθ
- ( '
Figure imgf000006_0001
利用阵列加权矢量对各通道传输信号进行幅度和相位调整， .从而将阵列的空域辐射 声束导向的期望的区域内。
进一步地， 歩骤 4 ) 中所述多比特 Δ调制， 其处理过程如下： 首先， 通过插 值滤波器， 将均衡处理后的高比特 PCM 编码按照指定的过采样因数进行插值滤波 处理， 获得过采样的 PCM编码信号； 然后， 进行∑- Δ调制处理， 将音频带宽范 I 内的噪声能量推挤到音频带之外， 保证了系统在音频带内具有足够高的信噪比， 同 时经∑- Δ调制处理后， 原高比特 PCM码变换为低比特 PCM码， 其 PCM编码的比 特位数得到了缩减。
进一步地， 步骤 4 ) 中所述多比特∑- Δ调制， 其所采用的 Σ - Δ调制方法， 是 按照现有各种∑ - Δ调制方法——像高阶单级（Higher~Order Single-Stage)串行调制 方法或者多级 （Multi-stage (Cascade, MASH) )并行调制方法， 对插值滤波器输出的 过采样信号进行噪声整形处理， 将噪声能量推挤到音频带之外， 保证了系统具有足 够高的带内信噪比。
进一步地， 歩骤 5) 中所述温度计编码转换， 用于将位宽为 M的低比特 PCM编 码信号转换为对应于 个通道的数字功放和换能器负载的一元码矢量。该一元码矢 量的每个数位上的编码会送到相应的数字通道上， 其各数位上的编码， 在任意时刻 仅有 "0"和 " 1 "两种电平状态， 在 "0"状态时， 换能器负载被关断， 在 " 1 "状 态时， 换能器负载被开通。 温度^编码操作， 用于将编码信息分配到多个换能器负 载通道， ^而将换能器负载纳入到信号编码流程中， 实现了对换能器阵列的数字化 编码和数字式开关控制。
进一步地， 步骤 6 ) 中所述动态失配整形处理， 用于对温度†编码矢量进行重 新排序， 进一步优化一元码矢量的数据分配方案， 消除由陈元之间频响差异引起的 空域合成信号中的非线性高次谐波失真分量。
进一歩地， 步骤 6 ) 中所述动态失配整形处理， 通过采用现有各种方式的整形 算法——像 DWA (Data-Weighted Averaging,数据加权平均)、 VFMS(Vector-Feedback mismatch-shaping 向量反馈失配整形 )和 TSMS(Tree-Structure mismatch-shaping 整形操作， 压低带内谐波失真成份的强度， 将其功率推挤到带外高频段， 认而降低 了带内的谐波失真强度， 提高了∑ - Δ编码信号的音质水平。
进一步地， 步骤 7 ) 中所述通道信息抽取， 对各通道抉行编码信息分配操作， 其信号处理过程如下： 如图 4所示， 首先， 各通道的动态失配器进行动态失配整形 处理， 经整形处理后， 获得了位序更新的整形矢量； 然后按照特定的袖取选择准则， 从每个通道的整形矢量的 2M个数位中， 选取一个指定的数位编码作为该通道的输 出编码， 为保证信息的完整还原， 各通道选取的数位位次不能存在重复， 旦所有的 2M个通道选取的数位位次完全包含了 1到 2M个数位位次。 在通道信息抽取选择过 程中， 一般可以按照第 i个通道.从其整形矢量中选取第 i个数位编码信息的简单准 则进行数位选取操作。 经过多个通道的比特位抽取选择及合并操作之后， 多个阵元 通道上的预先设置的均衡和波束加权处理操伤得到了有效的继承， 从而为数字化阵 列的均衡和指向性控制操作提供了一种有效的实现途径。
进一步地， 步骤 7 ) 中所述负载可以为多个扬声器单元组成的数字化扬声器阵 列， 也可以为具有多个音圈绕组的扬声器单元， 还可以为多个多音圈扬声器单元组 成的数字扬声器阵列。
本发明另一方面提供一种具有通道均衡和波束控制功能的数字化扬声器阵列系 统装置， 包括- 一音源， 是系统待播放的信息；
一数字转换器， 与音源的输出端相连接， 用于将输入信号转换为位宽为 N、 采 样率为 .；的高比特 PCM编码信号；
一通道均衡器， 与数字转换器的输出端相连接， 于对各通道频响进行逆滤波 均衡操伤， 消除通道频响的带内起伏；
一波束形成器， 与通道均衡器的输出端相连接， ffl于控制扬声器阵列波束的空 域辐射形状， 产生像 3D立体声场、虚拟环绕声场、指向性声场等声场分布特性， 从 而达到特殊声效播放的目的；
一 Δ调制器， 与波束形成器的输出端相连接，用于完成过采样插值滤波和多 比特∑ Δ编码调刺处理， 获得位宽缩减的低比特 PCM编码信号；
一温度计编码器，与 Σ - Δ调制器的输出端相连接，用干将低比特 PC 编码信号 转换为与系统数字通道数相等的一元码矢量,从而用于数字化通道开关的控制矢量； 一动态失配整形器， 与温度 ^编码器的输出端相连接， 用于消除由阵元之间频 响差异引入的空域合成信号的非线性谐波失真分量， 压低音频带内谐波失真成份的 强度，将这些谐频成份的功率推挤到带外高频段，从而降低了带內的谐波失真强度， 提高了∑ Δ编码信号的音质水平；
一抽取选择器， 与动态失配整形器的输出端相连接， 用于从各通道的整形矢量 中， 抽取特定的数位编码信息， 用于控制该通道进行开通 /关断动作的控刺信息； 一多通道数字功放， 与袖取选择器的输出端相连接， ]¾于将各通道的控制编码 信号进行功率放大， 用于驱动后级数字化负载进行开通 /关断；
一数字化阵列负载， 与多通道数字功放的输出端相连接， 用于完成电声转换操 作， 将数字化的幵关电信号转换为模拟格式的空气振动信号。
进一歩地， 音源可以为模拟信号或者数字编码信号， 可以来自于各种模拟装置 所产生的模拟音源信号， 也可以是各种数字装置所产生的数字编码信号。
进一歩地， 数字转换器， 可以包含模数转换器、 USB、 LAN, COM 等数字接 口电路和接口协议程序， 能够与现有的数字接口格式相兼容， 通过这些接口电路和 协议程序， 数字化扬声器阵列系统装置， 能够灵活方便的与其他装置设备进行信息 的交互与传递； 同时， 经过数字转换器处理后， 原来的输入的模拟或者数字音源信 号转换为位宽为 N , 采样率为 的高比特 PCM编码信号。
进一步地， 通道均衡器可以在时域或者频域内按照逆滤波的响应参数进行均衡 操作， 消除各通道音频带内的频响起伏； 同^， 也校正了各通道的频响差异， 使各 通道频响趋干一致。
进一步地， 波束形成器利用所设^的加权矢量， 对各通道传输信号进行加权处 理， 调整其幅度和相位信息， 从而使复杂环境下数字化阵列的空域方向图达到期望 的设 i卜要求。
进一歩地， ∑- Δ调制器， 其信号处理过程如下： 首先， 将原来位宽为 N、 采样 率为. /;的 PCM编码按过采样因子 。进行过采样的插值滤波处理， 获得位宽为 N、 采样率为？ 的 PCM编码信号；然后按照多比特 Σ Δ调制方式，将位宽为 N的过 采样 PCM编码信号转换成位宽为 M (M<N) 的低比特 PCM编码信号， 从而缩减了 PCM编码信号的位宽。
进一步地， ∑ Δ调制器， 可以按照现有各种 Σ Δ调制器的信号处理结构 像高阶单级串行调制器结构或者多级并行的调制器结构， 对插值滤波输出的过采样 信号进行噪声整形处理， 将噪声能量推挤到音频带之外， 保证了系统具有足够高的 带内信噪比。
进一歩地温度计编码器用干将位宽为 h4的低比特 PC 编码信号转换为对应于 2M个通道的数字功放和换能器负载的一元码信号矢量，该一元码矢量的每个数位编 码信息分配到一个对应的数字化通道上，从而将换能器负载纳入到信号编码流程中， 实现了对换能器负载的数字化编码和数字式开关控制。
进一步地，动态失配整形器，通过采用现有的各种整形算法 - 像 DWA, VTMS 和 TSMS算法， 将由阵元之间频响差异引入的非线性谐波失真频谱进行整形操作， 压低带内谐波失真成份的强度， 将其功率推挤到带外高频段， .从而降低了带内的谐 波失真强度， 提高了 Σ - Δ编码信号的音质水平。
进一歩地， 抽取选择器按照特定的抽取准则， 将 2 个数字通道的每个通道 整形矢量中， 袖取一个数位信息， 作为该通道的输出编码信息， 用于控制后级的换 能器负载进行开通 /关断动作。经过袖取选择器的比特袖取与合并操作之后，原来多 个通道的均衡器响应和通道指向性加权矢量的操作都得到了有效的实现， 保证了数 字化阵列的频响平坦性和波束方向的可控性。
进一步地， 多通道数字功放将抽取选择器输出的开关信号送至全桥式功放电路 的 MOSFET管栅极端， 通过控制 MOSFET管的导通与关断来控制功率电源到负载 供电的开通与关断， 从而实现了对数字负载的功率放大。
进一步地， 数字化阵列负载可以为多个扬声器单元组成的数字化阵列， 也可以 为多个音圈的扬声器单元， 还可以为多音圈扬声器组成的陈列。 数字化负载的每个 数字通道可以由单个或者多个扬声器单元组成； 也可以由单个或者多个音圈组成； 还可以由多个音圈和多个扬声器单元组合而成。 数字化负载的阵列形状， 可以根据 换能器单元数量和实际应 ]¾需求进行排列， 组成适合干实际应 需求的各种阵列形 状。
与现有技术相比， 本发明的优点在于： 实现了系统整个信号传输链路的全数字 化， 整个系统装置完全由数字化器件组成， 便于进行高度的集成化电路设 i卜， 提高 了系统的工作稳定性， 降低了系统的功耗、 体积和重量； 同时， 数字化扬声器阵列 系统， 能够灵活方便的与其他数字化系统设备进行数据交互， 能够更好的适应于数 字化的发展要求。多比特 Σ Δ调制技术通过噪声整形方法，将音频带内的噪声功率 推挤到带外高频区域， 而保证了音频带内的高信噪比要求， 这种调制技术的硬件 实现电路简单，对电路器件制作过程中所产生的参数偏差具有很好的免疫力。另外，
A. 本发明所采用的全数子化系统实现方式，其抗干扰能力更强，在复杂的电磁 干扰环境中能够保证稳定可靠的工作。
B. 本发明所釆用的动态失配整形算法，能够有效地消减因阵元之间频响差异引 入的非线性谐波失真强度， 提高系统的音质水平， 因此该系统对于换能器单元之间 的频响偏差具有很好的免疫力。
C.通过温度计编码方法， 给各换能器单元分配相应的一元码信号， 使得各扬声 器单元 （或者各音圈） 工作在开通或关断状态， 这种交替开关工作的状态， 有效地 避免了各扬声器单元 （或者各音圈） 出现过载失真现象， 从而延长了各扬声器单元 (或者各音圈） 的使用寿命； 同时， 换能器采用开关工作方式， 其电声转换效率更 高， 换能器的发热更少。
I 数字功放电路， 直接将放大后的开关信号送到扬声器端， 控制扬声器进行幵 通与关断操作， 不需要在数字功放后级加入体积较大、 份格昂贵的电感电容进行模 拟低通处理， 缩减了系统体积与成本； 同时， 对于呈容性特性的压电换能器负载来 讲， 通常需要加电感进行阻抗匹配， 以增加压电扬声器的输出声功率， 而在换能器 端施加数字信号 ， 其阻抗匹配效果要优于传统的在换能器端施加模拟信号的阻抗 匹配效果。
E. 温度计编码方式使得每组阵元所分配的一元码信号,仅包含原有音源信号的 部分信息成份， 单纯依赖单组阵元所辐射的信息不能完成音源信息的完整还原， 只 有联合所有分组阵元空域辐射声场的合成作用， 才能完整的还原出音源信息； 这种 联合多组阵元空域辐射声场的合成作用完成信息还原的工作方式， 其还原信息具有 空域指向性， 在阵列对称轴线上具有最大信噪比， 偏离轴线越远， 其信噪比越低。
H. 通道均衡方法能够保持各通道音频带内频响平坦，并校正了通道之间的频响 差异， 保证了系统还原音源信号频谱与原始音源信号的真实频谱趋干一致， 从而保 证了数字重放系统真实的再现原来音源的声场效果； 同时， 这种均衡方法所带来的 各通道音频带内频响平坦性和通道间频响一致性， 为各种自适应算法具有较好的稳 定性、 较快的收敛速度、 较好的鲁棒性提供了有利的支撑。
I. 基于数据抽取选择的通道均衡方法， 能够较好的抑制各通道的频响起伏， 提 高了数字化系统的声场还原质量， 并 ϋ能够消除通道之间较大的频响差异性， 因此 经过多通道均衡处理后， 通道间的频响偏差得到了较大程度的补偿， 仅剩下了少量 的残留偏差， 这些残留偏差能够进一步依靠失配整形算法进行较好的校正处理， 从 而将失配整形算法去除少量偏差的能力也得以有效发挥。 经通道均衡处理后， 阵元 的频响差异性得到了较好的校正， 从而保证了各种基于阵元通道相干积累的陈列波 束控制算法能够得以有效运行。这种基于数据抽取选择的数字化阵列波束形成方法， 能够有效提高数字化阵列在复杂环境下的空域声场控制能力。
.].波束控制方法， 保证了数字化扬声器阵列在复杂环境下具有较好的波束指向 性， 通过袖取选择的信息合并方式， 使得常规的波束控制方法， 可以很好的应用于 数字化阵列的波束控制， 为实 环境中特殊声场 （如 3D立体声场、 虚拟环绕声场、 指向性声场等） 效果的生成提供了有效的实现途径。 K.数据抽取选择方法， 能够将传统的基于 PCM编码格式的通道均衡和波束形 成算法， 直接扩展应用于基于多比特 Σ - Δ调制的数字化阵列系统中， 从而为传统通 道均衡和波束控制算法与基于多比特 Σ - Δ调刺的数字化阵列系统之间建立了桥梁， 保证了传统算法能够继续在基于多比特 Σ - Δ调制的阵列系统中发挥有效通道均衡 和波束导向作用。
附图说明
图 1表示根据本发明的一种具有通道均衡和波束控制功能的数字化扬声器系统 装置的各组成模块示意图；
图 2表示本发明在通道均衡参数估计过程中通道参数测量示意图；
图 3表示本发明在波束控制过程中的通道权矢量加载示意图；
图 4表示本发明在通道信息抽取过程中所采 ]¾的抽取规则示意图；
图 5表示本发明一实施例在通道均衡过程中所采 ]¾逆滤波器的幅度谱曲线图； 图 6表示本发明一实施例的 Σ Δ调制器所采用的 5阶 CiFB调制结构的信号处 理流程图；
图 Ί表示本发明一实施倒的温度计编码矢量的开光控制示意图；
图 8表示本发明一实施例的动态失配整形器所釆用的 VFMS失配整形算法的信 号处理流程图；
图 9表示本发明一实施例的袖取选择器所釆用的抽取准则示意图；
图 10表示本发明一实施飼的 8元扬声器阵列布放示意图；
图 1 1表示本发明一实施 ^的扬声器阵列和传声器単元的布放位置示意图； 图 12表示本发明一实施例的阵列轴线 1 米处位置点在均衡前后其系统频响的 幅度谱曲线对比图；
图 13 表示本发明一实施例在- ·60 度、 0 度和 +30 度三种预定方向上产生的波 束方向图曲线；
图 14表示本本发明一实施例的∑ Δ调制器所采用的参数值。
具体实施方式
下面结合 i 图和具体实施方式对本发明作进一步详细描述- 本发明首先通过数字转换接 将可听声范 内的音源信号转换成位宽为 N的 高比特 PCM 编码信号； 然后利用通道均衡技术， 对各通道的数字音源信号进行逆 滤波均衡处理， 消除各通道音频带内频响起伏， 同时消除通道间频响差异性； 然后 利 ^波束形成技术， 对均衡后的各通道信号进行加权处理， 使阵列能够导向到期望 的空间方向上；然后再利用多比特 Σ - Δ调制技术将位宽为 N的高比特 PCM编码信 号转换成为位宽为 M ( <N) 的低比特 PCM编码信号： 然后再通过温度计编码方 法将位宽为 M的 PCM编码信号转换为位宽为 2M的温度†编码， 形成分配到 2 组 换能器陈元的一元码信号； 然后再经过动态失配整形技术， 对分配到各组阵元的一 元码信号进行动态失配整形处理，消除因各组阵元频响差异所引入的高次谐波分量， 降低系统的总谐波失真， 提升系统的音质水平； 最后通过抽取选择技术， 从每个通 道的失配整形矢量中， 抽取一个数位上的比特信息， 送至该通道的数字功放形成功 率信号， 驱动该通道的数字化负载进行开通或关断操作， 所有通道的数字化负载所 辐射的空域声场进行叠加后在空间某预定区域内还原出源信号。
如图 1所示,制作一个依据本发明的具有通道均衡和波束控制功能的数字化扬 声器系统装置， 其主体由音源 1、 数字转换器 2、 通道均衡器 3、 波束形成器 4、 ∑~ Δ调制器 5、 温度^编码器 6、 动态失配整形器 7、 抽取选择器 8、 多通道数字功放 9以及数字化阵列负载 10等组成。
音源 1， 可以选用在 PC机硬盘内存储的 MP3格式的音源文件， 可以通过 USB 端口按数字格式输出： 也可以选用 MP3播放器内存储的音源文件，通过模拟格式输 出； 还可以利用信号源产生音频范围内的测试信号， 也通过模拟格式输出。
数字转换器 2， 与所述音源 1 的输出端连接， 包含数字输入格式和模拟输入格 式两种输入接口， 针对数字输入格式， 采用 Ti公司的一款型号为 PCM2706的 USB 接口芯片， 能够将 PC机内存储的 MP3类型文件经由 USB端口按照】6比特位宽、 44.1 KHz采样率通过 I2S接口协议实时读入到型号为 Cyclone III EP3C80F484C8的 FPGA芯 j†内； 针对模拟输入格式，采用 Analog Devices公司的一款型号为 AD1877 的模数转换芯片， 将模拟音源信号转换为 i6比特、 44.1 KHz的 PCM编码信号， 也 通过 I2S接口协议实时读入到 FPGA芯片内。
通道均衡器 3, 与所述数字转换器 2的输出端相连接， 按照测量方式， 计算出 各通道的逆滤波器参数， 图 5给出了通道 1到 8的逆滤波器幅度谱曲线， 按照逆滤 波器参数对各通道进行均衡处理， 获得均衡后的 16 比特、 44. 1 KHz采样率的 PCM ί Ρ !J。
波束形成器 4, 与所述通道均衡器 3的输出端相连接， 按照期望的波束方向图 计算 8元阵列的权值矢量，然后在 FPGA内部，通过乘法器单元将计算的权值矢量加 载到各阵元通道的传输信号——均衡后的 16比特、 44. 1 KHz采样率的 PCM信号， 从而形成带有方向加权调整的多通道 PC 信号。
Σ - Δ调制器 5， 与所述波束形成器 4的输出端相连接， 首先， 在 FPGA芯片内 部， 进行过采样的插值滤波操作， 将 44.:Ι ΚΉζ、 16比特的 PCM编码信号， 按三级 进行升采样插值处理， 第一级插值因子为 4， 采样率升为 Π6,4ΚΗζ, 第二级插值因 子为 4, 采样率升为 705.6 ΚΉζ, 第 Ξ:级插值因子为 2， 采样率升为 1411.2 ΚΗΖ„ 在 经过 32倍插值处理后， 原 44.1 ΚΉζ、 16 比特的 PCM信号转换为 1,41】2 MHz, 】6 比特的过采样 PCM 信号； 然后按照 3 比特的∑- Δ调制方式， 将过釆样的 1.4112 MHz, 16 比特的 PCIV [编码信号转换成为 1.4112 MHz > 3比特的 PCMb编码信号。 如图 6所示， 在本实施例中， ∑- Δ调制器采用 5阶 CIFB Cascaded Integrators with Distributed Feedback)的拓扑结构。该调制器的系数如表 1所示。为了节约硬件资源， 降低其实现代份， 在 FPGA芯片内部， 通常会采用移位加法运算来代替常数乘法运 算， 并将 Σ-Δ调制器所使用的参数用 CSD编码表示。
温度^编码器 6， 与所述 Σ Δ调制器 5的输出端相连接， 将 1,4112 MHz 、 3 比特的 Σ-Δ调制信号按照温度计编码方式转换为 1.4112 MHz, 位宽为 8的一元码。 如图 7所示， 当 3比特 PCM编码为 "001", 其转换的温度计编码为 " 00000001 该编码用于控制换能器阵列的 1个阵元开通， 其余 7个阵元都关闭； 当 3比特 PCM 编码为 "100" 时， 其转换的温度计编码为 "00001111 ", 该编码用于控制换能器陈 列的 4个阵元开通， 其余 4个陈元关闭； 当 3比特 PCM编码为 "111"， 其转换的 温度计编码为 "01111111", 该编码用于控制换能器阵列的 7个阵元开通， 仅留下 1 个阵元关闭。
动态失配整形器 7， 与温度计编码器 6的输出端相连接， 用于消除阵元之间频 响差异引起的非线性谐波失真分量。 动态失配整形器 7按照非线性谐波失真分量最 少的优化准则， 对 8位温度 if编码进行排序， 从而决定出给 8个换能器阵元的编码 分配方式， 如图 7所示， 当温度 if编码为 "OOOOUil" ， 通过动态失配整形器进行 次序排列后， 将决定换能器阵元 1、 4、 5、 7上分配编码 " 1" , 换能器阵元 2、 3、 6、 8上分配编码 "0" , 丛而按照这一分配方式， 换能器阵元 1、 4、 5、 7将开通而 换能器阵元 2、 3、 6、 8将关闭，按照这一编码分配方式进行换能器阵列的开关控制， 将会使陈列辐射声场所合成的信号中包含最少的谐波失真分量。 在本实施例中， 动 态失配整形器采用了 VFMS算法， 其信号处理流程如图 8所示，其中粗线代表 V维 矢量， 细线代表标量，输入信号 V是经 Σ-Δ调制器和温度计编码器处理后的 W维编 码矢量， 该编码矢量中包含 V个 "】 "状态和^""^个 " 0"状态， 输出信号 SV是经 过失配整形处理后的 N维列矢量， 通过失配整形处理， 输出矢量的 "1"状态和 "0" 状态在矢量中的排列顺序得到了调整， 但是 "1"状态和 "0"状态的数量仍保持 不变， 并旦矢量中的每个元素控制着阵列中相应的一个阵元通道按照其状态进行通 断操作。 单元选择模块通过某种选择策略保证由频响差异引入的误差在频谱上能够 得到较好的整形效果， - min()模块表示在 N维矢量中选取数值最小的元素同时对 其取负， 经 - minG模块操作所获得的标量元素为 u, mtf是失配整形函数， 其一般 形式为 1 ， M为阶数， 本实施例中所采用的失配整形器阶数为 2阶。按照图 8的信号处理流图， 可以获得失配整形处理后的输出矢量表达式为- sv = u[l 1 … 1 ],χ ιΥ + mif (se ) ,
其中 se=sv~~y。 假设 v维行矢量 表示阵列各单元之间的不一致误差， ϋ假设 中所有元素之和为 0 , 则扬声器阵列在空间任意位置点上由各阵元输出声场进行叠 加合成后所获得的阵列输出声信号表达式为：
X - sv X ed
- [u[l 1 … ΐ]1κΛΓ (se)jx ed
Figure imgf000014_0001
= u x Q + mtf( )x eA 阵列输出声信号的表达式可以看出， mff 整形函数能够对阵列误差 进行整形 处理， 只要选择较好的失配整形函数 mtf, 就可以取得对阵列误差 的较好整形效 果。 在 FPGA芯片内部， 通过动态失配整形器处理后， 原 Σ - Δ编码信号中存在的谐 波分量被推挤到带外高频段， .从而提高了带内音源信号的音质水平。
抽取选择器 8， 与动态失配整形器 7的输出端相连接， 用于 各通道的整形矢 量中进行数位抽取操作， 送给后级功放和数字负载。 如图 9所示， 每个通道经失配 整形处理都产生了一个 8元的一元码矢量， 抽取选择器 7将会按照第 i个通道抽取 整形矢量第 个数位的原則，为每个通道抽取一个相应数位的一元码信号，作为后级 数字功放的输入信号。
多通道数字功放 9， 与袖取选择器 8的输出端相连接。 本实施例中， 数字功放 芯片选用 Ti公司的一款型号为 TAS512〗的数字功放芯片，该芯片的响应时间在 100 ns量级， 能够无失真响应 1.4112 MHz的一元码流信号。 在功放的输入端， 采用差 分输入格式， 在 FPGA内部， 将动态失配整形送来的输出数据一路直接输出， 另一 路经反相后输出， 形成了两路差分信号， 送到 TAS5121芯片的差分输入端； 在功放 的输出端， 同样采用差分输出格式， 将两路差分信号直接施加到单个换能器阵元通 道的正负极引线上。
数字化陈列负载 〗0, 与多通道数字功放 9的输出端相连接。本实施例中， 数字 化负载单元采用惠威公司生产的型号为 B2S的全频带扬声器单元，该单元的频带范 围为 270 Hz〜20 KHz,灵敏度 (2,83V/lni)为 79 (IB,最大功率为 2 W,额定阻抗为 8 欧 姆。如图 10所示， 数字化负载为 8元扬声器阵列， 该陈列由 8个上述扬声器单元按 照线性阵列方式摆放， 阵元间距为 4 cm, 每个扬声器单元对应一个数字化通道。
在自由空间中， 假设扬声器阵列和传声器单元的布放如图 ] 1 所示， 按照仿真 实验方法， 假设给数字化扬声器系统装置输入频率范围为 100 Hz〜20 KHz的扫频信 号，在扬声器阵列轴线上 1米远位置点处观察系统的频响特性。图 12给出了在均衡 器施加前后， 轴线 1米远位置点处系统频响的幅度谱曲线对比图， 在未施加均衡器 时， 系统频响的幅度谱在 2 KHz〜20 KHz的频率范 i簡内存在着非常明显的下降趋势， 随着频率.从 2 KHz增加到 20 KHz, 系统频响的幅度谱从 65 dB下降到 45 dB， 存在 着 20 dB的幅度差异； 在施加均衡器之后， 系统频响的幅度谱在 2 KHz〜20 KHz的 频率范围内一直维持在 57 dB附近， 呈现出非常平坦的频谱特性， 从而保证了系统 合成信号的真实还原。 根据均衡结果可知， 采用抽取选择的多通道比特信息合成方 式， 能够有效的继承各通道的均衡器响应信息， 保证了各通道的频响平坦性。
基于通道均衡的数字化扬声器阵列系统， 能够有效地消除各通道声频带内的频 响起伏， 并校正通道之间的频响差异， 保证了系统在期望的空间区域内具有非常平 坦的时域频响特性， 从而保证了所有通道在空间合成信号的频谱能够还原出原始音 源信号的真实频谱， 保证了数字重放系统真实再现原来音源的声场效果。 另外， 通 过消除各通道音频带内频响起伏也保证了各种自适应空域阵列波束形成算法具有较 快的收敛速度和较好的鲁棒性。
在自由空间中， 仍然按图 i l 所示的扬声器阵列布放方式， 按照- 60 度、 0 度 和 +30 度三种预定的波束主瓣方向， 进行阵列波束控制的仿真实验， 设定三种情况 的阵列波瓣宽度都为 20度。 图 13给出了三种预定方向情况 T阵列的空域方向图， 观察这些曲线可以看出， 阵列的波束主瓣指向到预定方向， 波束宽度达到了期望的 要求， 主副瓣幅度差值达到了 15 dB, 根据这些阵列波束控制结果可知， 采 抽取 选择的多通道信息合成方式， 能够有效的继承波束形成器所加载在各通道上的幅度 和相位调整信息， 认而实现了阵列的波束指向性控制。 这种基于抽取选择方式的数 字化阵列波束形成方法,能够有效提高数字化阵列在复杂环境下的空域指向性能力， 为数字化阵列特殊声场 （如 3D立体声场、 虛拟环绕声场、 指向性声场等） 效果的 生成提供了可靠的实现途径。
以上实施例仅用以说明本发明的技术方案而非限制。 尽管参照实施例对本发明 进行了详细说明， 本领域的普通技术人员应当理解， 对本发明的技术方案进行修改 或者等同替换， 均应涵盖在本发明的权利要求范围当中。

Claims

权禾 il要求；
1. 一种数字化扬声器阵列系统的通道均衡与波束控制方法， 包括如 Τ步骤：
1 ) 数字格式转换,将信号转换为基于 PCM编码的数字信号；
2) 通道均衡处理；
3) 控制波束形成；
4) 多比特∑- Δ调制;
5)温度计编码转换，将位宽为 Μ的低比特 PCM编码信号转换为对应于 2Μ个通 道的数字功放和换能器负载的一元码矢量；
6) 动态失配整形处理， 对温度计编码矢量进行重新排序；
7) 抽取通道信息， 送至数字功放驱动负载发声。
2. 根据权利要求 1所述的方法， 其中， 当步骤 1 ) 的数字格式转换针对模拟信号 时包括： 首先经过模数转换操作， 转换为基干 PC 编码的数字信号， 然后按照指定 位宽和采样率的参数要求进行变换， 转换为满足参数要求的 PC 编码信号。
3. 根据权利要求 1所述的方法， 其中， 当歩骤 1 ) 的数字格式转换针对数字信号时 包括： 按照指定位宽和采样率的参数要求进行变换， 转换为满足参数要求的 PCM编 码信号。
4. 根据权利要求 ]—所述的方法， 其中， 步骤 2) 的通道均衡处理， 其均衡器的参 数按照测量和 ^算获取。
5. 根据权利要求 i所述的方法， 其中， 步骤 3) 的控制波束形成， 其波束形成器的 通道权系数按照常规波束形成的设计方法进行计算， 计算公式为式 （ I ):
w ^ argmin ^ w a!
式 （ I )
Figure imgf000016_0001
其中， a ( 9 )为空域的导向矢量， α {θ ) ·■· ， Ν表示阵 列的阵元数量， D ( 9 )为期望的空域波束形状，
Figure imgf000016_0002
6. 根据权利要求 1所述的方法， 其中， 步骤 4) 的多比特∑- Δ调制， 其处理过程 如下： 首先， 通过插值滤波器， 将均衡处理后的高比特 PCM编码按照指定的过采 样因数进行插值滤波处理， 获得过采样的 PCM编码信号； 然后， 进行∑- Δ调制处 理， 将音频带宽范围内的噪声能量推挤到音频带之外， 原高比特 PCM码变换为低 比特 PCM码。
7. 根据权利要求 1所述的方法， 其中， 步骤 4) 的多比特 Σ-Δ调制采用高阶单级 串行调制方法或者多级并行调制方法， 对插值滤波器输出的过采样信号进行噪声整 形处理， 将噪声能量推挤到音频带之外。
8. 根据杈利要求 1所述的方法， 其中， 步骤 5) 中所述的一元码矢量的每个数位上 的编码送到相应的数字通道上， 其各数位上的编码， 在任意时刻有 "0"和 "1"两 种电平状态， 在 " 0"状态时， 换能器负载被关断， 在 " 1"状态时， 换能器负载被 开通。
9. 根据权利要求 1所述的方法， 其中， 步骤 6)的动态失配整形处理， 采用 DWA、 VFMS和 /或 TSMS整形算法，将由阵元之间频响差异引入的非线性谐波失真频谱进 行整形操作， 压低带內谐波失真成份的强度， 将其功率推挤到带外高频段。
10. 根据权利要求 1所述的方法， 其中， 步骤 7) 的通道信息抽取， 对各通道执行 编码信息分配操作， 其信号处理过程如下： 首先， 各通道的动态失配器进行动态失 配整形处理， 经整形处理后， 获得了位序更新的整形矢量； 然后按照特定的抽取选 择准剣， 认每个通道的整形矢量的 2M个数位中， 选取一个指定的数位编码作为该 通道的输出编码， 为保证信息的完整还原， 各通道选取的数位位次不能存在重复， ϋ所有的 2Μ个通道选取的数位位次完全包含了 1到 4个数位位次。
11. 根据权利要求 10 所述的方法， 其中， 在通道信息袖取选择过程中， 按照第 i 个通道从其整形矢量中选取第 i个数位编码信息的简阜准则进行数位选取操作。
12. 根据权利要求 1所述的方法， 其中， 步骤 7) 中所述的负载为多个扬声器单元 组成的数字化扬声器阵列， 或者为具有多个音圈绕组的扬声器单元， 或者为多个多 音圈扬声器单元组成的数字扬声器阵列。
13. 一种具有通道均衡和波束控制功能的数字化扬声器阵列系统装置，其特征在于 包括：
一音源 （1), 是系统待播放的信息；
一数字转换器（2)， 其与所述的音源（1) 的输出端相连接， 用于将输入信号转 换为位宽为 N、 采样率为 的高比特 PCM编码信号：
一通道均衡器（3)， 其与所述的数字转换器（2) 的输出端相连接， 用于对各通 道频响进行逆滤波均衡操作， 消除通道频响的带内起伏；
一波束形成器（4), 其与所述的通道均衡器（3) 的输出端相连接， ]¾于控制扬 声器阵列波束的空域辐射形状， 产生如 3D立体声场、虚拟环绕声场、指向性声场等 声场分布特性， 以达到特殊声效播放的目的；
一∑- Δ调制器 （5)， 其与所述的波束形成器 (4) 的输出端相连接， 用于完成 过采样插值滤波和多比特 Σ -Δ编码调制处理， 获得位宽缩减的低比特 PCM 编码信 号；
一温度计编码器 （6), 其与所述的 Δ调制器 （5) 的输出端相连接， 用于将 低比特 PCM编码信号转换为与系统数字通道数相等的一元码矢量， 以用于数字化通 道开关的控制矢量；
一动态失配整形器 (7), 其与所述的温度计编码器（6) 的输出端相连接， 用于 消除由阵元之间频响差异引入的空域合成信号的非线性谐波失真分量， 压低音频带 内谐波失真成份的强度， 将这些谐频成份的功率推挤到带外高频段， 从而降低了带 内的谐波失真强度， 提高了 Σ-Δ编码信号的音质水平；
一抽取选择器（8), 其与所述的动态失配整形器（7) 的输出端相连接， 于从 各通道的整形矢量中，抽取特定的数位编码信息，用于控制该通道进行开通 /关断动 作的控制信息；
一多通道数字功放（9)， 其与所述的抽取选择器（8) 的输出端相连接， 用于将 各通道的控制编码信号进行功率放大， 用于驱动后级数字化负载进行开通 /关断； 一数字化阵列负载 （10), 其与所述的多通道数字功放 （9) 的输出端相连接， 用于完成电声转换， 将数字化的开关电信号转换为模拟格式的空气振动信号。
14. 根据杈利要求 13所述的装置，其中，音源（1)为模拟信号或者数字编码信号。
15. 根据权利要求 13所述的装置， 其中， 数字转换器（2)包含模数转换器、 USB、 LAN, COM等数字接口电路和接口协议程序。
16. 根据权利要求 13所述的装置， 其中， 通道均衡器 3) 在 域或者频域内按照 逆滤波的响应参数进行均衡操作， 消除各通道音频带内的频响起伏， 并校正各通道 的频响差异。
17. 根据权利要求 13所述的装置， 其中， 波束形成器（4)利 K所设计的加权矢量， 对各通道传输信号进行加权处理， 调整其幅度和相位信息。
18. 根据权利要求 13所述的装置， 其中， ∑ Δ调制器 (5) 的信号处理过程如下； 首先， 将原来位宽为 V, 采样率为.；的 PCM编码按过采样因子 ^进行过采样的插 值滤波处理， 获得位宽为 N、 采样率为^ ：的: PCM编码信号： 然后按照多比特∑ Δ调制方式， 将位宽为 V的过釆样 PCM编码信号转换成位宽为 M ( <N)的低比特 PCM编码信号。
19. 根据权利要求 13所述的装置， 其中， ∑ Δ调制器 （5)， 按照高阶单级串行调 制器结构或者多级并行的调制器结构， 对插值滤波输出的过采样信号进行噪声整形 处理， 将噪声能量推挤到音频带之外。
20. 根据权利要求 13所述的装置， 其中， 温度†编码器（6 )将位宽为 M的低比特 PCtvi编码信号转换为对应于 2M个通道的数字功放和换能器负载的一元码信号矢量， 该一元码矢量的每个数位编码信息分配到一个对应的数字化通道上， 以将换能器负 载纳入到信号编码流程中， 对换能器负载的数字化编码和数字式开关控制。
21. 根据权利要求 13所述的装置， 其中， 动态失配整形器 （7 )采用 DWA、 VFMS 和 /或 TSMS整形算法，将由阵元之间频响差异引入的非线性谐波失真频谱进行整形 操作， 压低带内谐波失真成份的强度， 将其功率推挤到带外高频段， 降低带内的谐 波失真强度。
22. 根据权利要求 13所述的装置， 其中， 抽取选择器 8 ) 按照特定的抽取准则， 将从 2 个数字通道的每个通道整形矢量中， 抽取一个数位信息， 作为该通道的输 出编码信息， 用于控制后级的换能器负载进行开通 /关断动作。
23. 根据权利要求 13所述的装置， 其中， 多通道数字功放 （9 ) 将抽取选择器 （8 ) 输出的开关信号送至全桥式功放电路的 M0SFET管樋极端， 通过控制 ) SFET管的导 通与关断来控制功率电源到负载供电的开通与关断。
24. 根据权利要求 13所述的装置， 其中， 数字化阵列负载 （10 ) , 为多个扬声器单 元组成的数字化阵列， 其每个数字通道由单个或者多个扬声器单元组成； 或者为多 个音圈的扬声器单元， 其每个数字通道由单个或者多个音圈组成； 或者为多音圈扬 声器组成的阵列其每个数字通道由多个音圈和多个扬声器单元组合而成。
25. 根据权利要求 13或 24所述的装置， 其中， 数字化阵列负载（10 )的阵列形状， 根据换能器单元数量和实际应 需求进行排列。
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