CN102780964B - Professional loudspeaker system method of adjustment - Google Patents
Professional loudspeaker system method of adjustment Download PDFInfo
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Abstract
The present invention relates to speaker control technology, specifically a kind of speaker system regulation method.The method includes speaker system model establishment step, and virtual input processing parameter modification step, wherein this speaker system model includes some sound box model.Each sound box model is corresponding with an entity active audio amplifier, and including virtual input processing parameter and virtual output processing parameter, this virtual input processing parameter includes one or more virtual DSP data, and each DSP data include the DSP parameter that multilamellar kind is identical.The most no matter each entity DSP of active audio amplifier comprises how many layers of DSP parameter, and the input processing parameter multilamellar that this technology can realize system level is adjustable, thus simplifies the debugging correct operation of speaker system, improves debugging efficiency, and safeguards system debugging effect.
Description
Technical field
The present invention relates to speaker control technology, specifically a kind of professional loudspeaker system method of adjustment.
Background technology
The audio amplifier (also referred to as active audio amplifier) of embedded digital signal processor (DSP) and power amplifier can carry out the operations such as EQ adjustment, time delay, amplitude limit and frequency dividing by digital signal processor to input audio signal, makes audio amplifier reach optimum Working.
As follows to the conventional method of adjustment of the sound system of active audio amplifier composition at present: (1) first adjusts input processing parameter and output processing parameter (the above audio amplifier of two divided-frequency), the such as EQ, time delay, amplitude limit etc. of the digital signal processor of single audio amplifier；(2) effect of sound system is tested, refine test data by rule of thumb and further and the digital signal processor parameter of relevant audio amplifier is adjusted；(3) again the whole structure of sound system is tested, according to test result, again the digital signal processor parameter of active audio amplifier is adjusted.It is to say, for realizing preferably system acoustic effect, need, constantly according to overall acoustic efficiency, repeatedly the parameter of the digital signal processor of each audio amplifier to be adjusted.
The debugging bearing calibration of this sound system (speaker system) there is problems in that
(1) system debug correct operation is loaded down with trivial details, complicated, and workload is big.A lot of owing to affecting the factor of system acoustic effect, adjust which audio amplifier, and adjust which parameter of audio amplifier, there is no a fixed pattern, can only engineering staff can only audio amplifier or region-by-region (audio amplifier in region is also to be adjusted one by one) be adjusted one by one, again each audio amplifier is adjusted further according to test result after adjustment, i.e. needs be repeated continuously the acoustic efficiency of measurement system and adjust each audio amplifier, can be only achieved preferably system sound effect.The original workload that audio amplifier was adjusted one by one is the biggest, and also needs to now to repeat many times that (number of times is unknown, and spent time and required workload are the hugest.For layman, the task that sound system debugging correction has been practically impossible to.
(2) system is even across repeatedly adjusting correction, but still it cannot be guaranteed that can obtain preferably acoustic efficiency.As it has been described above, the factor affecting acoustic efficiency is a lot, many times, for time-consuming, engineering staff the most rule of thumb judges which should adjust which parameter of which audio amplifier.Even if engineering staff's audio amplifier one by one is adjusted, but owing to sound system is a systematic engineering of business, can influence each other between each audio amplifier, and engineering staff is being adjusted each audio amplifier when, it is difficult to prediction and judges the audio amplifier after adjusting, other audio amplifiers can be produced what impact (or what impact other audio amplifiers can produce to this audio amplifier), therefore can only estimate by rule of thumb.Therefore, this debugging bearing calibration, much less ordinary person, even for the personnel through professional training, also cannot ensure that sound can obtain gratifying system sound effect.
Therefore simplify sound system audio amplifier debugging correct operation, improve debugging efficiency, and safeguards system debugging effect is a technical problem urgently to be resolved hurrily.Solve this technical barrier, for the development and application promoting Specialty Hi-Fi technology, there is important function.
Summary of the invention
Present invention solves the technical problem that and be to provide a kind of professional loudspeaker system method of adjustment, to simplify the debugging correct operation of speaker system, improve debugging efficiency, and safeguards system debugging effect.
For solving above-mentioned technical problem, the technical solution used in the present invention is:
A kind of professional loudspeaker system method of adjustment, this speaker system includes multiple active audio amplifier, and this speaker system is provided with one or more array group, and each array group includes one or more active audio amplifier；Each active audio amplifier includes signal input unit, digital signal processor, power amplifier unit and loudspeaker unit；
This signal input unit is for receiving the audio signal of the outside input of audio amplifier, and transmits to this digital signal processor；
This digital signal processor is provided with data memory module, in this data memory module, storage has one group of input processing parameter, this group input processing parameter includes multiple entity DSP data, each entity DSP packet contains an audio amplifier layer DSP parameter corresponding with the kind of this entity DSP data, and this audio amplifier layer DSP parameter is the DSP parameter arranged for this active audio amplifier；And at least entity DSP data also include an array layer DSP parameter identical with its audio amplifier layer DSP parameter kind, this array layer DSP parameter is each audio amplifier member for array group belonging to this active audio amplifier and the DSP parameter that arranges；
This digital signal processor also includes input processing module, this input processing module includes the sound effect processor of multiple series connection, each sound effect processor respectively with each DSP parameter one_to_one corresponding of this group input processing parameter, each sound effect processor is for carrying out corresponding signal processing operations according to corresponding DSP parameter to audio signal；
This power amplifier unit is for for receiving the audio signal processed through this input processing module, and transmits to this loudspeaker unit after being amplified this audio signal processing；
This method of adjustment comprises the following steps:
Step S10: set up speaker system model, this speaker system model includes multiple sound box model, each sound box model is corresponding with an active audio amplifier in speaker system, each sound box model includes one group of virtual input processing parameter, this group virtual input processing parameter includes multiple virtual DSP data, each virtual DSP data include an audio amplifier layer DSP parameter and an array layer DSP parameter, each virtual DSP data of sound box model respectively with each entity DSP data one_to_one corresponding of active audio amplifier；
Step S20: if selecting to repair audio amplifier layer DSP parameter, perform step S21；If selecting amendment array layer DSP parameter, perform step S22；
Step S21: the audio amplifier layer DSP parameter that amendment is selected, and amended data are preserved to sound box model corresponding audio amplifier layer DSP parameter belonging to this audio amplifier layer DSP parameter；Perform step S24；
Step S22: the array layer DSP parameter that amendment is selected, if active audio amplifier corresponding to sound box model belonging to this array layer DSP parameter is not belonging to the audio amplifier member of arbitrary array group, if now this array layer DSP parameter can be modified, the most amended data will preserve to this sound box model corresponding array layer DSP parameter, perform step S24；If active audio amplifier corresponding to sound box model belonging to this array layer DSP parameter belongs to the audio amplifier member of a certain array group, then amended data will preserve to the array layer DSP parameter of each sound box model corresponding to this array group respectively, perform step S24；
Step S24: for the sound box model of virtual DSP data generation amendment:
If entity DSP packet corresponding to these virtual DSP data is containing audio amplifier layer DSP parameter, array layer DSP parameter, then in the corresponding DSP parameter in the DSP parameter synchronization changed in these virtual DSP data to correspondent entity DSP data；
If entity DSP data corresponding to these virtual DSP data only comprise audio amplifier layer DSP parameter, the most each layer DSP parameter superposition calculation of these virtual DSP data is obtained a new DSP parameter, and by the audio amplifier layer DSP parameter of this new DSP parameter synchronization to correspondent entity DSP data；
Perform step S20 or step S30；
Step S30: terminate.
Compared with prior art, provide the benefit that:
The sound effect of whole sound system is if desired optimized by prior art, need each audio amplifier is adjusted the most repeatedly correction, but also belonging to this audio amplifier, how the acoustics of array group and whole sound system can change after the unpredictable indivedual audio amplifier of adjustment, add difficulty and the complexity of system debug correction, therefore to debug not only difficulty bigger for the sound system of prior art, very professional sound technique personnel are needed to operate, and the workload debugged is very big, and debugging effect also cannot ensure.
Active sound box control method based on this technology, engineers and technicians can give up the traditional method accurately adjusting each audio amplifier, then by the basis of each audio amplifier of basic correction, in units of group, the audio amplifier member to each array group unifies debugging correction, the acoustics (transmission frequency characteristic) making each array group and whole sound system reaches the use requirement of system, optimize the overall acoustic efficiency of sound system to greatest extent, but the difficulty of sound system debugging and workload are greatly reduced.
This technology thinks deeply the debugging correction work of whole audio amplifier from the angle that system uses, both the self-characteristic of audio amplifier had been taken into full account, take into full account again between each audio amplifier of same array group, interactional factor between each array group of same system, by single audio amplifier DSP parameter adjustment, the parameter adjustment of array group DSP, from audio amplifier itself, the DSP parameter of each audio amplifier is adjusted by two aspects of speaker array group, this adjustment seems the most accurate, it is not that each audio amplifier can be debugged to optimum state, but but can obtain optimal system sound effect this just sound system actually used required.
Sum it up, this technology is possible not only to simplify the difficulty of sound system debugging correction, reduces workload, but also the system sound effect making us more being satisfied with can be obtained.
Accompanying drawing explanation
The structural representation of the active audio amplifier of Fig. 1 embodiment；
The digital signal processor architecture schematic diagram of the active audio amplifier of Fig. 2 embodiment；
The principle schematic of the digital signal processor of the active audio amplifier of Fig. 3 embodiment；
Fig. 4 is the flow chart of the speaker system regulation method of embodiment.
Detailed description of the invention
Below in conjunction with the accompanying drawings embodiments of the invention are further described.
The speaker system of the professional loudspeaker system method of adjustment of the present embodiment includes multiple active audio amplifier, and this speaker system is provided with one or more array group, and each array group includes one or more active audio amplifier.The present embodiment uses active audio amplifier as described below.
As it is shown in figure 1, this active audio amplifier includes signal input unit 11, digital signal processor 12, power amplifier unit 13 and loudspeaker unit 14.This signal input unit 11 is for receiving the audio signal of the outside input of audio amplifier, and transmits to this digital signal processor 12.As in figure 2 it is shown, this digital signal processor 12 is provided with data memory module 124, in this data memory module 124, storage has one group of input processing parameter, and this group input processing parameter includes multiple entity DSP data.The kind of entity DSP data does not limit, and input processing parameter can comprise the entity DSP data of multiple identical type, it is also possible to comprise different multiple entity DSP data.
Each entity DSP packet contains an audio amplifier layer DSP parameter corresponding with the kind of this entity DSP data, this audio amplifier layer DSP parameter is the DSP parameter arranged for this active audio amplifier, and at least entity DSP data also include an array layer DSP parameter identical with its audio amplifier layer DSP parameter kind.This array layer DSP parameter is each audio amplifier member for array group belonging to this active audio amplifier and the DSP parameter that arranges.
For sound box model, audio amplifier layer DSP parameter is the DSP parameter arranged for single sound box model, array layer DSP parameter is then the DSP parameter arranged for each sound box model associated with same array group, the DSP parameter that each sound box model i.e. associated with same array group is shared.The kind of each (each layer) DSP parameter that entity DSP packet contains is identical, assume that entity EQ packet is containing audio amplifier layer EQ parameter and array layer EQ parameter, so the DSP parameter kind of this audio amplifier layer EQ parameter and this array layer EQ parameter is all EQ parameter, the corresponding sound effect processor of each EQ parameter.
As shown in Figure 2, this digital signal processor 12 also includes input processing module 121, this input processing module 121 includes the sound effect processor of multiple series connection, each sound effect processor respectively with each DSP parameter one_to_one corresponding of this group input processing parameter, each sound effect processor is for carrying out corresponding signal processing operations according to corresponding DSP parameter to audio signal.
This power amplifier unit 13 is for for receiving the audio signal processed through this input processing module 121, and transmits to this loudspeaker unit 14 after being amplified this audio signal processing.
In this technique, one DSP parameter is not meant to this DSP parameter only one of which numerical value, one DSP parameter is also likely to be and is made up of multiple (one group the most groups) numerical value, digital signal processor 12 performs corresponding signal processing (audio effect processing) operation according to a DSP parameter, to change the transmission frequency characteristic of audio amplifier.The most here " a DSP parameter ", it can be appreciated that be the one or more data required for certain signal processing operations, such as muting parameter and gain parameter have only to data, and EQ parameter then needs one group of data.In digital signal processor 12, a corresponding sound effect processor of DSP parameter.
Each sound effect processor of input processing module 121 is series connection, and series connection (execution) order of each sound effect processor can be determined by any-mode, it is also possible to is determined by some way.Wherein, the audio signal handled by sound effect processor made number one is directly from signal input unit 11；The follow-up audio signal handled by each sound effect processor processes through previous sound effect processor；Come a last sound effect processor and also need to the audio signal transmission after being processed to power amplifier unit 13.
In this technique, for adjust single active audio amplifier DSP parameter (can be any type of DSP parameter) be referred to as audio amplifier layer DSP parameter, the i.e. data of this audio amplifier layer DSP parameter be this active audio amplifier exclusive, with other audio amplifiers share.This active audio amplifier can be adjusted by the audio amplifier layer DSP parameter of amendment active audio amplifier.
And the DSP parameter (being probably any type of DSP parameter) being used for adjusting the audio amplifier member characteristics of array group is referred to as array layer DSP parameter simultaneously, each audio amplifier member of the most same array group shares the data of this array layer DSP parameter.In array group, certain array layer DSP parameter of some audio amplifier member is modified, and the corresponding array layer DSP parameter of remaining audio amplifier member in the most same array group can be also identical numerical value by synchronous vacations.The characteristic of each active audio amplifier belonging to same array group can be adjusted, it is achieved the synchronization control of multiple audio amplifiers by the array layer DSP parameter of amendment active audio amplifier simultaneously.
Audio amplifier layer DSP parameter is relative concept with array layer DSP parameter, in the prior art, only has audio amplifier layer DSP parameter, therefore when system call interception, can only be adjusted by audio amplifier one by one in the input processing parameter of active audio amplifier.And this technology is by arranging the array group DSP parameter of association in each audio amplifier member of array group, if revising the array group DSP parameter of one of them audio amplifier member, the array group DSP parameter that so other audio amplifiers member of this array group is corresponding all can be by synchronous vacations (needing by controlling platform or control system realization), thus realize the adjustment of the synchronization of multiple audio amplifier, simplify difficulty and complexity that speaker system adjusts.
Therefore this technology is possible not only to realize the adjustment correction of single audio amplifier, the audio to each audio amplifier group and speaker system are overall can also be realized and be adjusted correction, thus simplify the adjustment correction work of whole sound system, and ensure to ensure last adjustment calibration result.
With reference to Fig. 3, the input processing parameter of the active audio amplifier of the present embodiment includes following one or more entity DSP data: input quiet data, input gain data, input time delay data, input reversed polarity data, input air attenuation compensation data, input compression sliced data, EQ data；
The sound effect processor that each DSP parameter of the entity DSP data that input processing module 121 comprises with input processing parameter is comprised of digital signal processor 12 is corresponding:
The each input muting parameter comprised for this input quiet data, corresponding sound effect processor is the quiet sound effect processor of input for audio signal carries out silence switch process according to this input muting parameter；
The each input gain parameter comprised for these input gain data, corresponding sound effect processor is the input gain sound effect processor for audio signal carries out input gain process according to this input gain parameter；
The each input time delay parameter comprised for these input time delay data, corresponding sound effect processor is the input time delay sound effect processor for audio signal carries out input time delay process according to this input time delay parameter；
The each input reversed polarity parameter comprised for these input reversed polarity data, corresponding sound effect processor is the input reversed polarity sound effect processor for audio signal carries out reversed polarity operation process according to this input reversed polarity parameter；
The each input air attenuation compensation parameter comprised for these input air attenuation compensation data, corresponding sound effect processor is the input air attenuation compensation sound effect processor for audio signal carries out attenuation of air compensation deals according to this input air attenuation compensation parameter；
The each input compression clipping parameter comprised for this input compression sliced data, corresponding sound effect processor is the input compression amplitude limit sound effect processor for audio signal is compressed amplitude limiting processing according to this input compression clipping parameter；
For each EQ parameter, corresponding sound effect processor is the EQ sound effect processor (EQ sound effect processor can use IIR second order filter) for audio signal is compressed amplitude limiting processing according to this EQ parameter；
Each entity DSP data of the present embodiment comprise the audio amplifier layer DSP parameter corresponding with its kind, such as input quiet data comprises audio amplifier layer input muting parameter, input gain packet layer input gain parameter Han audio amplifier, input time delay packet layer input time delay parameter Han audio amplifier, input reversed polarity packet inputs reversed polarity parameter containing audio amplifier layer, input air attenuation compensation packet layer input air attenuation compensation parameter Han audio amplifier, input compression sliced data comprises audio amplifier layer input compression clipping parameter, EQ packet layer EQ parameter Han audio amplifier.At least one of which entity DSP data, in addition to comprising audio amplifier layer DSP parameter, also comprise array layer DSP parameter, say, that part entity DSP data only comprise monolayer parameter, and part entity DSP packet contains multilamellar DSP parameter；Or all DSP data all comprise multilamellar DSP parameter.
The value volume and range of product of the sound effect processor that active audio amplifier input processing module 121 is comprised, be by its value volume and range of product of the actual DSP parameter comprised determined.Such as, if input processing parameter only comprises a muting parameter and an input time delay parameter, then input processing module 121 will only comprise a quiet sound effect processor and an input time delay sound effect processor；If input processing parameter comprises 2 muting parameter and 2 input time delay parameters, then input processing module 121 will comprise 2 quiet sound effect processors and 2 input time delay sound effect processors.
The sound effect processor comprised due to input processing module 121 is series connection, and the order of connection of each sound effect processor can determine according to any-mode, it is also possible to determines according to ad hoc fashion.Wherein, the handled audio signal of first sound effect processor is from signal input unit 11；Audio signal handled by follow-up sound effect processor is from a upper sound effect processor；Last sound effect processor also needs to the audio signal transmission after processing to corresponding power amplifier unit 13.
As shown in Figure 4, the professional loudspeaker system method of adjustment of the present embodiment comprises the following steps:
Step S10: set up speaker system model, this speaker system model includes multiple sound box model, each sound box model and the active audio amplifier corresponding (the most corresponding) in speaker system, each sound box model includes one group of virtual input processing parameter, this group virtual input processing parameter includes multiple virtual DSP data, each virtual DSP data include an audio amplifier layer DSP parameter and an array layer DSP parameter, each virtual DSP data of sound box model respectively with each entity DSP data one_to_one corresponding of active audio amplifier；
Step S20: if selecting to repair audio amplifier layer DSP parameter, perform step S21；If selecting amendment array layer DSP parameter, perform step S32；
Step S21: the audio amplifier layer DSP parameter that amendment is selected, and amended data are preserved to sound box model corresponding audio amplifier layer DSP parameter belonging to this audio amplifier layer DSP parameter；Perform step S24；
Step S22: the array layer DSP parameter that amendment is selected, if active audio amplifier corresponding to sound box model belonging to this array layer DSP parameter is not belonging to the audio amplifier member of arbitrary array group, if now this array layer DSP parameter can be modified (i.e. can set the array layer DSP parameter of the sound box model not associated with array group as closing or opening), the most amended data will preserve to this sound box model corresponding array layer DSP parameter, perform step S24；If active audio amplifier corresponding to sound box model belonging to this array layer DSP parameter belongs to the audio amplifier member of a certain array group, then amended data will preserve to the array layer DSP parameter of each sound box model corresponding to this array group respectively, perform step S24；(each sound box model is with each active audio amplifier one to one, then each self-corresponding sound box model of each audio amplifier member of array group and this array group are to have corresponding relation)
Step S24: for the sound box model of virtual DSP data generation amendment:
If entity DSP packet corresponding to these virtual DSP data is containing audio amplifier layer DSP parameter, array layer DSP parameter, then the DSP parameter (being probably audio amplifier layer DSP parameter or array layer DSP parameter number) changed in these virtual DSP data is synchronized in the corresponding DSP parameter in correspondent entity DSP data；
If entity DSP data corresponding to these virtual DSP data only comprise audio amplifier layer DSP parameter, each layer DSP parameter superposition calculation of these virtual DSP data the most first obtains a new DSP parameter, and (the superposition calculation operation of each layer DSP parameter can pass through audio amplifier internal resource, the most built-in microprocessor 15 completes, can also be by the resource outside audio amplifier, such as sound system controls platform to be completed), and by the audio amplifier layer DSP parameter of this new DSP parameter synchronization to correspondent entity DSP data；
Perform step S20 or step S30；
Step S30: terminate.
In the present embodiment, no matter which floor (several) DSP parameter is each entity DSP data of active audio amplifier input processing parameter specifically comprise, and each virtual DSP data of the sound box model corresponding with this active audio amplifier comprise multilamellar DSP parameter the most respectively.Each entity DSP data that each the virtual DSP data comprised due to sound box model and active audio amplifier are comprised be one_to_one corresponding.Active audio amplifier the actual entity DSP data comprised, sound box model all exists the virtual DSP data corresponding with this entity DSP.
', sound box model B ' and sound box model C in the present embodiment, for step S30: assuming that array group first includes active audio amplifier A, active audio amplifier B and active audio amplifier C, the sound box model of its correspondence is sound box model A respectively '.The input processing parameter of each active audio amplifier includes entity EQ data, and the virtual input processing parameter of the most each sound box model includes virtual EQ data, and these virtual EQ data include an audio amplifier layer EQ parameter and an array layer EQ parameter.
(1) amendment audio amplifier layer DSP parameter
If in the audio amplifier layer EQ parameter of amendment sound box model A ' audio amplifier layer EQ parameter, then amended data will be saved in sound box model A '.Due to sound box model A ' the virtual EQ data of virtual input processing parameter are modified, it is therefore desirable to this amendment is synchronized to the active audio amplifier A of correspondence.
Now, if the entity EQ data of active audio amplifier contain 2 layers of EQ parameter: audio amplifier layer EQ parameter and array layer EQ parameter, then sound box model A ' audio amplifier layer EQ parameter by the audio amplifier layer EQ parameter being synchronized to active audio amplifier A.
Now, if the entity EQ data of active audio amplifier only comprise audio amplifier layer EQ parameter, so it being accomplished by first by this sound box model A ' audio amplifier layer EQ parameter in virtual EQ data, array layer EQ parameter superposition calculation obtain new EQ parameter, and the superposition audio that obtains in active audio amplifier A of two EQ parameters before superposition to obtain audio with the EQ parameter of superposition calculation gained in active audio amplifier A consistent, in the audio amplifier layer EQ data of the data transfer of the newest EQ parameter to corresponding active audio amplifier.
(2) amendment array layer DSP parameter
If amendment sound box model A ' array layer EQ parameter, due to sound box model A ' corresponding to active audio amplifier A belong to the audio amplifier member of array group first, the most amended data will preserve to the array layer EQ parameter of each sound box model corresponding to array group first respectively, and the most amended data will preserve respectively to sound box model A ', sound box model B ', sound box model C ' array layer EQ parameter in.Although here, only have adjusted sound box model A ' array layer EQ parameter, but with this sound box model A ' associate other two sound box model B ' and sound box model C ' array layer EQ parameter all there occurs amendment.', sound box model B ', sound box model C it is thus desirable to by sound box model A ' data syn-chronization revised is in each self-corresponding active audio amplifier.
Now, ', sound box model B ' and sound box model C if the entity EQ data of active audio amplifier contain audio amplifier layer EQ parameter, array layer EQ parameter, then sound box model A ' array layer EQ parameter be synchronized to respectively in the array layer EQ parameter of the most corresponding active audio amplifier.
Now, if the entity EQ data of active audio amplifier only comprise audio amplifier layer EQ parameter, superposition calculation is obtained new EQ parameter, and the new EQ parameter each obtained is synchronized to respectively in the audio amplifier layer EQ parameter of corresponding active audio amplifier by the so audio amplifier layer EQ parameter of each sound box model, array layer EQ parameter.By sound box model B ' as a example by, first its audio amplifier layer EQ parameter, array layer EQ parameter superposition calculation are obtained new EQ parameter, the most again by the audio amplifier layer EQ parameter of this new EQ parameter synchronization to active audio amplifier B.
Multiple DSP parameters of same DSP parameter respectively process after superposition audio, and first these DSP parameters are overlapped calculate after carry out processing obtained audio for parameter stack result again, the audio result that the two processing mode obtains is the same.Each entity DSP data such as sporocarp audio amplifier input processing parameter comprise multilamellar, then every kind of DSP parameter is required for processing repeatedly, so can increase the weight of the burden of signal processor, limits its processing speed.Assume that the input processing parameter of certain active audio amplifier includes quiet data, delay data, gain data and reversed polarity data, and each entity DSP data include 2 layers of DSP parameter respectively: audio amplifier layer DSP parameter and array layer DSP parameter, so digital signal processor 12 of this active audio amplifier needs to perform twice silence switch process, twice delay process, twice gain process and twice reversed polarity process, the input processing module 121 of digital signal processor 12 includes 2 quiet sound effect processors in other words, 2 time delay sound effect processors, 2 gain sound effect processors and 2 reversed polarity sound effect processors.
For solving the problems referred to above, reduce the audio effect processing number of times of active audio amplifier digital signal processor 12, improve treatment effeciency and processing speed.In the present embodiment, even if the part entity DSP data of active audio amplifier input processing parameter are arranged to monolayer, but each virtual DSP data of sound box model fictionalize multilamellar DSP parameter in system model.For the entity DSP data of only one layer of DSP parameter in active audio amplifier, after corresponding virtual DSP data modification, first each layer DSP parameter of these virtual DSP data is superimposed as a DSP parameter, then resynchronize to this entity DSP data, thus realize the multilamellar adjustable effect of each DSP data of system level.
The present embodiment can be by the resource beyond digital signal processor 12, such as can control to realize the professional loudspeaker system method of adjustment described in this technology on platform at sound box system, control the sound box model of platform fictionalizes multilamellar DSP parameter, then when controlling platform and amended DSP supplemental characteristic being synchronized to entity audio amplifier, first the microprocessor 15 in controlling platform or active audio amplifier completes corresponding DSP parameter overlap-add operation, reduce the data volume managed and the amount of calculation of digital signal processor 12, raising audio effect processing efficiency of making rational use of resources.Therefore the present embodiment can make full use of the external resource of digital signal processor 12, reduces audio effect processing quantity (or number of times), improves the processing speed of digital signal processor 12.
In the present embodiment, active audio amplifier also includes microprocessor 15, and this microprocessor 15 is for being adjusted each DSP parameter of this group input processing parameter according to external control signal.
Accordingly, in step s 24:
If entity DSP packet corresponding to these virtual DSP data is containing audio amplifier layer DSP parameter, array layer DSP parameter, first send the control signal of amendment corresponding (corresponding with these virtual DSP data revised) entity DSP data according to microprocessor 15 to corresponding (corresponding with this sound box model) active audio amplifier in the virtual DSP data being modified, the corresponding DSP parameter in the entity DSP data to correspondence of the DSP parameter in these virtual DSP data that this microprocessor 15 is comprised according to this control signal is modified；
If entity DSP data corresponding to these virtual DSP data only comprise audio amplifier layer DSP parameter, first according to these virtual DSP data send the control signal of amendment correspondent entity DSP data to the microprocessor 15 of corresponding active audio amplifier, each layer DSP parameter superposition calculation of these virtual DSP data that this this control signal is comprised by this microprocessor 15 obtains a new DSP parameter, then modifies the audio amplifier layer DSP parameter of corresponding entity DSP data further according to this new DSP parameter；
Additionally, this active audio amplifier also includes the control signal interface being connected with microprocessor 15, this control signal interface is used for receiving external control signal and transmitting to microprocessor 15.
This technology is when implementing, for the DSP parameter that some is relatively simple, and the most quiet, time delay, reversed polarity, gain etc., the input processing parameter of active audio amplifier is provided only with one layer of DSP parameter.Carrying out system debug timing, each layer DSP parameter of the virtual DSP data corresponding with these entities DSP data first passes through external resource (resource beyond digital signal processor 12) and is overlapped calculating one DSP parameter of synthesis, in the most corresponding DSP data of the most corresponding active audio amplifier of transmission.And for complex DSP parameter, such as EQ, the input processing parameter of active audio amplifier arranges 2 layers of EQ parameter, is audio amplifier layer EQ parameter, array layer EQ parameter respectively.If certain layer of EQ parameter of sound box model is revised, then in one layer of (one) EQ parameter that the EQ parameter of amendment is corresponding by being directly synchronized to active audio amplifier, and without first carrying out the superposition calculation of multiple DSP parameter.Thus both made full use of the efficient fast signal disposal ability of digital signal processor 12, external resource can be made full use of again, improve the real-time audio signal disposal ability of digital signal processor 12 further.
Claims (5)
1. a professional loudspeaker system method of adjustment, it is characterised in that: this speaker system includes multiple active audio amplifier, and this speaker system is provided with one or more array group, and each array group includes one or more active audio amplifier；Each active audio amplifier includes signal input unit, digital signal processor, power amplifier unit and loudspeaker unit；
This signal input unit is for receiving the audio signal of the outside input of audio amplifier, and transmits to this digital signal processor；
This digital signal processor is provided with data memory module, in this data memory module, storage has one group of input processing parameter, this group input processing parameter includes multiple entity DSP data, each entity DSP packet contains an audio amplifier layer DSP parameter corresponding with the kind of this entity DSP data, and this audio amplifier layer DSP parameter is the DSP parameter arranged for this active audio amplifier；And at least entity DSP data also include an array layer DSP parameter identical with its audio amplifier layer DSP parameter kind, this array layer DSP parameter is each audio amplifier member for array group belonging to this active audio amplifier and the DSP parameter that arranges；
This digital signal processor also includes input processing module, this input processing module includes the sound effect processor of multiple series connection, each sound effect processor respectively with each DSP parameter one_to_one corresponding of this group input processing parameter, each sound effect processor is for carrying out corresponding signal processing operations according to corresponding DSP parameter to audio signal；
This power amplifier unit is for receiving the audio signal processed through this input processing module, and transmits to this loudspeaker unit after being amplified this audio signal processing；
This method of adjustment comprises the following steps:
Step S10: set up speaker system model, this speaker system model includes multiple sound box model, each sound box model is corresponding with an active audio amplifier in speaker system, each sound box model includes one group of virtual input processing parameter, this group virtual input processing parameter includes multiple virtual DSP data, each virtual DSP data include an audio amplifier layer DSP parameter and an array layer DSP parameter, each virtual DSP data of sound box model respectively with each entity DSP data one_to_one corresponding of active audio amplifier；
Step S20: if selecting amendment audio amplifier layer DSP parameter, perform step S21；If selecting amendment array layer DSP parameter, perform step S22；
Step S21: the audio amplifier layer DSP parameter that amendment is selected, and amended data are preserved to sound box model corresponding audio amplifier layer DSP parameter belonging to this audio amplifier layer DSP parameter；Perform step S24；
Step S22: the array layer DSP parameter that amendment is selected, if active audio amplifier corresponding to sound box model belonging to this array layer DSP parameter is not belonging to the audio amplifier member of arbitrary array group, if now this array layer DSP parameter can be modified, the most amended data will preserve to this sound box model corresponding array layer DSP parameter, perform step S24；If active audio amplifier corresponding to sound box model belonging to this array layer DSP parameter belongs to the audio amplifier member of a certain array group, then amended data will preserve to the array layer DSP parameter of each sound box model corresponding to this array group respectively, perform step S24；
Step S24: for the sound box model of virtual DSP data generation amendment:
If entity DSP packet corresponding to these virtual DSP data is containing audio amplifier layer DSP parameter, array layer DSP parameter, then in the corresponding DSP parameter in the DSP parameter synchronization changed in these virtual DSP data to correspondent entity DSP data；
If entity DSP data corresponding to these virtual DSP data only comprise audio amplifier layer DSP parameter, the most each layer DSP parameter superposition calculation of these virtual DSP data is obtained a new DSP parameter, and by the audio amplifier layer DSP parameter of this new DSP parameter synchronization to correspondent entity DSP data；
Perform step S20 or step S30；
Step S30: terminate.
A kind of professional loudspeaker system method of adjustment the most according to claim 1, it is characterized in that: the described input processing parameter of described active audio amplifier includes following multiple entity DSP data: input quiet data, input gain data, input time delay data, input reversed polarity data, input air attenuation compensation data, input compression sliced data, EQ data；
The sound effect processor that each DSP parameter of the entity DSP data that the input processing module of described digital signal processor comprises with described input processing parameter is comprised is corresponding:
The each input muting parameter comprised for this input quiet data, corresponding sound effect processor is the quiet sound effect processor of input for audio signal carries out silence switch process according to this input muting parameter；
The each input gain parameter comprised for these input gain data, corresponding sound effect processor is the input gain sound effect processor for audio signal carries out input gain process according to this input gain parameter；
The each input time delay parameter comprised for these input time delay data, corresponding sound effect processor is the input time delay sound effect processor for audio signal carries out input time delay process according to this input time delay parameter；
The each input reversed polarity parameter comprised for these input reversed polarity data, corresponding sound effect processor is the input reversed polarity sound effect processor for audio signal carries out reversed polarity operation process according to this input reversed polarity parameter；
The each input air attenuation compensation parameter comprised for these input air attenuation compensation data, corresponding sound effect processor is the input air attenuation compensation sound effect processor for audio signal carries out attenuation of air compensation deals according to this input air attenuation compensation parameter；
The each input compression clipping parameter comprised for this input compression sliced data, corresponding sound effect processor is the input compression amplitude limit sound effect processor for audio signal is compressed amplitude limiting processing according to this input compression clipping parameter；
For each EQ parameter, corresponding sound effect processor is the EQ sound effect processor for audio signal is compressed amplitude limiting processing according to this EQ parameter.
Professional loudspeaker system method of adjustment the most according to claim 1 and 2, it is characterised in that: described active audio amplifier also includes microprocessor, and this microprocessor is for being adjusted each DSP parameter of this group input processing parameter according to external control signal.
Professional loudspeaker system method of adjustment the most according to claim 3, it is characterised in that:
In described step S24,
If entity DSP packet corresponding to these virtual DSP data is containing audio amplifier layer DSP parameter, array layer DSP parameter, first send the control signal of amendment correspondent entity DSP data to the microprocessor of corresponding active audio amplifier according to the virtual DSP data being modified, the corresponding DSP parameter in corresponding entity DSP data is modified by the DSP parameter in these virtual DSP data that this microprocessor is comprised according to this control signal；
If entity DSP data corresponding to these virtual DSP data only comprise audio amplifier layer DSP parameter, first send the control signal of amendment correspondent entity DSP data to the microprocessor of corresponding active audio amplifier according to these virtual DSP data, each layer DSP parameter superposition calculation of these virtual DSP data that this this control signal is comprised by this microprocessor obtains a new DSP parameter, then modifies the audio amplifier layer DSP parameter of corresponding entity DSP data further according to this new DSP parameter.
Professional loudspeaker system method of adjustment the most according to claim 3, it is characterised in that: this active audio amplifier also includes the control signal interface being connected with described microprocessor, and this control signal interface is used for receiving external control signal and transmitting to described microprocessor.
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| CN103916774A (en) * | 2012-12-31 | 2014-07-09 | 广州励丰文化科技股份有限公司 | Remote control method of active integrated loudspeaker based on multi-DSP system |
| CN104460631A (en) * | 2014-12-15 | 2015-03-25 | 浙江大丰实业股份有限公司 | Distributed type stage sound effect unified allocation system |
| CN106375926B (en) * | 2016-12-08 | 2019-04-02 | 深圳市信维声学科技有限公司 | Side is spoken the test method and its system of speaker |
| CN106488377B (en) * | 2016-12-08 | 2019-04-02 | 深圳市信维声学科技有限公司 | It is just speaking the test method and its system of speaker |
| CN108200526B (en) * | 2017-12-29 | 2020-09-22 | 广州励丰文化科技股份有限公司 | Sound debugging method and device based on reliability curve |
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