CN109151667B - Signal processing method, device and loudspeaker - Google Patents
Signal processing method, device and loudspeaker Download PDFInfo
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- CN109151667B CN109151667B CN201811105556.1A CN201811105556A CN109151667B CN 109151667 B CN109151667 B CN 109151667B CN 201811105556 A CN201811105556 A CN 201811105556A CN 109151667 B CN109151667 B CN 109151667B
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Classifications
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- 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
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2420/00—Details of connection covered by H04R, not provided for in its groups
- H04R2420/03—Connection circuits to selectively connect loudspeakers or headphones to amplifiers
Abstract
The invention discloses a signal processing method, a device and a loudspeaker, which comprise the steps of filtering an input signal of a sound source to obtain a low-frequency signal and a high-frequency signal, wherein the low-frequency signal is subjected to gain and delay to obtain a gain output signal, the high-frequency signal is subjected to dynamic range control, delay and loudness control to obtain a loudness output signal, and finally the gain output signal and the loudness output signal are combined, and finally a required audio signal is output. The audio signal processing method disclosed by the invention can limit the displacement of the loudspeaker diaphragm and improve the overall volume.
Description
Technical Field
The invention relates to the technical field of information, in particular to a signal processing method, a signal processing device and a loudspeaker.
Background
The loudspeaker is a conversion device for converting electric energy into sound, when different electronic energy is transmitted to the coil, the coil generates energy to interact with the magnetic field of the magnet, the interaction causes vibration of the paper disc, the coil of the loudspeaker moves forwards or backwards, and due to the change of the electronic energy, the paper disc of the loudspeaker moves along with the change of the electronic energy, and the actions cause the density degree of air to change to generate sound.
Because, loudspeaker can lead to the fault rate of loudspeaker to increase and life to appear reducing when driving the device of air sound production through the vibrations of inside diaphragm. In the prior art, the vibration amplitude of the loudspeaker diaphragm is limited by performing gain control on an input sound source signal and by the sound source signal after gain control.
However, the sound source signal output by the gain control according to the related art causes a problem that the overall volume of the speaker is affected.
Disclosure of Invention
In view of the above, the embodiments of the present invention provide a signal processing method, a signal processing device, and a speaker, so as to solve the problem that the overall volume of the speaker is affected after an input sound source signal is processed in the prior art.
In order to achieve the above object, the embodiment of the present invention provides the following technical solutions:
a signal processing method, comprising:
filtering the sound source input signal Vin to obtain a low-frequency signal Vi1 and a high-frequency signal Vi2;
determining a gain signal based on the low-frequency signal Vi1, and delaying the gain signal according to preset time to obtain a gain output signal Vo1;
controlling the high-frequency signal Vi2 based on a dynamic range control algorithm, determining a loudness signal, and delaying the loudness signal according to preset time to obtain a loudness output signal Vo2;
and combining the gain output signal Vo1 and the loudness output signal Vo2 to obtain a sound source output signal Vout.
Preferably, the determining a gain signal based on the low frequency signal Vi1 and delaying the gain signal according to a preset time to obtain a gain output signal Vo1 includes:
obtaining diaphragm displacement M of a predicted sound source input signal according to a loudspeaker model;
judging whether the sound source input signal needs gain according to the diaphragm displacement M, and obtaining a corresponding gain signal according to a judging result;
delaying the sound source input signal according to preset time to obtain a delay signal;
and processing the delay signal based on the gain signal to obtain a gain output signal Vo1.
Preferably, the method for controlling the high frequency signal Vi2 based on the dynamic range control algorithm, determining a loudness signal, and delaying the loudness signal according to a preset time to obtain a loudness output signal Vo2 includes:
controlling the dynamic range of the high-frequency signal Vi2 according to a dynamic range control algorithm, and determining a loudness signal meeting the preset loudness requirement;
delaying the audio input signal according to preset time to obtain a delay signal;
and processing the delay signal based on the loudness control to obtain a loudness output signal Vo2.
Preferably, filtering the audio input signal Vin to obtain a low frequency signal Vi1 and a high frequency signal Vi2 includes:
performing high-pass filtering processing on the sound source input signal Vin to obtain a high-frequency signal Vi2;
and performing low-pass filtering processing on the sound source input signal Vin to obtain a low-frequency signal Vi1.
The second aspect of the present invention discloses a signal processing apparatus comprising: the device comprises a filtering module, a high-frequency signal processing module, a low-frequency signal processing module and a signal synthesizing module;
the filtering module is used for filtering the sound source input signal Vin to obtain a low-frequency signal Vi1 and a high-frequency signal Vi2, transmitting the low-frequency signal Vi1 to the low-frequency signal processing module and transmitting the high-frequency signal Vi2 to the high-frequency signal processing module;
the low-frequency signal processing module is used for determining a gain signal based on the low-frequency signal Vi1, delaying the gain signal according to preset time to obtain a gain output signal Vo1, and inputting the gain output signal Vo1 as one end of the signal synthesizing module;
the high-frequency signal processing module is used for controlling the high-frequency signal Vi2 based on a dynamic range control algorithm, determining a loudness signal, delaying the loudness signal according to preset time to obtain a loudness output signal Vo2, and inputting the loudness output signal Vo2 as the other end of the signal synthesizer;
the signal synthesizer is configured to combine the gain output signal Vo1 and the loudness output signal Vo2 to obtain a sound source output signal Vout.
Preferably, the high frequency signal processing module includes: first multiplier, parallel dynamic range control module and first delay device
The dynamic range control module is used for controlling the dynamic range of the high-frequency signal Vi2 according to a dynamic range control algorithm, determining a loudness signal Vo2 meeting the preset loudness requirement, and taking the loudness signal Vo2 as one input of the multiplier;
the first delayer is used for delaying the input signal according to preset time to obtain a delay signal, and taking the delay signal as the other input of the multiplier;
and the first multiplier is used for processing the delay signal based on the loudness signal Vo2 to obtain a loudness output signal Vo2.
Preferably, the low frequency signal processing module includes: the second multiplier is a first branch composed of a second delayer, and a second branch composed of a loudspeaker model and a gain calculation module which are connected in series;
the loudspeaker model is used for predicting the vibration film displacement M of the sound source input signal and transmitting the vibration film displacement M to the gain calculation module;
the gain calculation module is configured to determine whether the sound source input signal Vin needs gain according to the diaphragm displacement M, obtain a corresponding gain signal according to a determination result, and use the gain signal as an input of the second multiplier;
the second delayer is configured to delay the input signal Vi1 according to a preset time, obtain a delay signal Vi1, and use the delay signal as another input of the second multiplier;
the second multiplier is configured to process the delay signal based on the gain signal to obtain a gain output signal Vo1.
Preferably, the filtering module includes: a high pass filter and a low pass filter connected in parallel;
the high-pass filter is used for performing high-pass filtering processing on the sound source input signal Vin to obtain a high-frequency signal Vi2, and taking the high-frequency signal Vi2 as the input of the high-frequency signal processing module;
the low-frequency filter is configured to perform low-pass filtering processing on the audio input signal Vin to obtain a low-frequency signal Vi1, and take the low-frequency signal Vi1 as an input of the low-frequency signal processing module.
Preferably, the signal synthesis module includes an adder.
A third aspect of the invention discloses a loudspeaker comprising: the signal processing device disclosed in the second aspect of the invention comprises a digital-to-analog converter, an audio power amplifier module and a loudspeaker;
the input end of the digital-analog converter is connected with the output end of the signal processing device, and the output end of the digital-analog converter is connected with the input end of the audio power amplifier module;
the output end of the audio power amplifier module is connected with the input end of the loudspeaker.
According to the embodiment of the invention, the low-frequency signal Vi1 and the high-frequency signal Vi2 are obtained by filtering the sound source input signal Vin, wherein the gain and the delay of the low-frequency signal Vi1 can be carried out to obtain the gain output signal Vo1, the loudness output signal Vo2 can be obtained by dynamic range control, delay and loudness control on the high-frequency signal Vi2, the obtained gain output signal Vo1 and the loudness output signal Vo2 are combined, and finally the signal Vout is output. The processing method disclosed by the embodiment of the invention can effectively improve the overall volume of the loudspeaker, so that the signal processing method disclosed by the embodiment of the invention can limit the displacement of the loudspeaker diaphragm and improve the overall volume.
Drawings
In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the drawings that are required to be used in the embodiments or the description of the prior art will be briefly described below, and it is obvious that the drawings in the following description are only embodiments of the present invention, and that other drawings can be obtained according to the provided drawings without inventive effort for a person skilled in the art.
Fig. 1 is a flowchart of a signal processing method according to a first embodiment of the present invention;
fig. 2 is a flowchart of another signal processing method according to a first embodiment of the present invention;
fig. 3 is a graph showing a frequency characteristic of a speaker displacement according to a first embodiment of the present invention;
FIG. 4 is a flowchart of another signal processing method according to a first embodiment of the present invention;
fig. 5 is a schematic structural diagram of a signal processing device according to a second embodiment of the present invention;
fig. 6 is a schematic structural diagram of a signal processing device according to a second embodiment of the present invention;
fig. 7 is a schematic diagram of a speaker structure according to a third embodiment of the present invention.
Detailed Description
The following description of the embodiments of the present invention will be made clearly and completely with reference to the accompanying drawings, in which it is apparent that the embodiments described are only some embodiments of the present invention, but not all embodiments. All other embodiments, which can be made by those skilled in the art based on the embodiments of the invention without making any inventive effort, are intended to be within the scope of the invention.
In this application, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or apparatus that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or apparatus. Without further limitation, an element defined by the phrase "comprising one … …" does not exclude the presence of other like elements in a process, method, article, or apparatus that comprises the element.
Example 1
Referring to fig. 1, a flowchart of a signal processing method according to an embodiment of the present invention is shown, including the following steps:
step S101: the audio input signal Vin is filtered to obtain a low-frequency signal Vi1 and a high-frequency signal Vi2.
It should be noted that the frequency range of the sound effect signal is 20Hz to 20KHz.
In the embodiment of the invention, the frequency of the high-frequency signal is defined as between 2KHz and 20KHz, and the frequency of the low-frequency signal is defined as between 20Hz and 2 KHz.
When step S101 is performed, the following processing means are included:
performing high-pass filtering processing on the sound source input signal Vin to obtain a high-frequency signal Vi2;
and performing low-pass filtering processing on the sound source input signal Vin to obtain a low-frequency signal Vi1.
The above-mentioned high-pass filtering and low-pass filtering processes for the sound source input signal Vin do not limit the order of execution of the two in the embodiment of the present invention. Typically performed simultaneously.
Step S102: and determining a gain signal based on the low-frequency signal Vi1, and delaying the gain signal according to a preset time to obtain a gain output signal Vo1.
In executing step S102, referring to fig. 2, the method specifically includes the following steps:
step S201: and obtaining the diaphragm displacement M of the input signal of the predicted sound source according to the loudspeaker model.
The speaker model is built by a speaker displacement amplitude-frequency characteristic curve. The specific speaker displacement amplitude-frequency characteristic is shown in fig. 3. The basic process of building the speaker model is as follows:
first, an excitation signal is added across the speaker.
Alternatively, the excitation signal may be white noise or a swept sine wave signal.
Then, the diaphragm displacement of the speaker and the voltage signal are measured and recorded simultaneously.
Finally, fitting the speaker model parameters.
Optionally, the fitting of the speaker model parameters uses a least squares method.
Step S202: judging whether the sound source input signal needs gain according to the diaphragm displacement M, and obtaining a corresponding gain signal according to a judging result.
In the step S202, the specific process is to detect the frequency maximum value of the low-frequency signal within a period of time, calculate the required gain value according to the currently detected maximum value, and when the maximum value is greater than a certain amplitude, the gain is reduced, which is basically inversely proportional to the signal amplitude.
Step S203: and delaying the sound source input signal according to a preset time to obtain a delay signal. It should be noted that, the time delay time for the audio input signal, that is, the preset time, is the same as the time taken to perform the step S201 and the step S202 to finally obtain the gain signal.
Step S204: and processing the delay signal based on the gain signal to obtain a gain output signal Vo1.
It should be noted that, in the process of executing step S204, a multiplier may be used to multiply the gain signal with the delay signal, so as to obtain the gain output signal Vo1.
Step S103: and controlling the high-frequency signal Vi2 based on a dynamic range control algorithm, determining a loudness signal, and delaying the loudness signal according to preset time to obtain a loudness output signal Vo2.
In step S103, the duration of the delay of the loudness signal according to the preset time, that is, the preset time, is the same as the time taken to perform step S102 to finally obtain the gain output signal Vo1.
That is, in the signal processing method disclosed in the embodiment of the present invention, the gain output signal Vo1 is obtained to be output in time synchronization with the loudness output signal Vo2.
In executing step S103, referring to fig. 4, the following steps are specifically included;
step S401: and controlling the dynamic range of the high-frequency signal Vi2 according to a dynamic range control algorithm, and determining a loudness signal meeting the preset loudness requirement.
It should be noted that, the dynamic range control is a standard algorithm for audio processing, and is used to control the ratio of the peak value to the average value of the signal. While music with a small dynamic range has small sound fluctuation but larger loudness, music with a large dynamic range generally has large sound fluctuation but smaller loudness.
That is, the dynamic range corresponds to the loudness in such a way that the smaller the dynamic range, the smaller the loudness.
Step S402: and delaying the audio input signal according to a preset time to obtain a delay signal.
It should be noted that, the time delay time for the audio input signal, that is, the preset time, is the same as the time taken to obtain the loudness signal in step S401.
Step S403: and processing the delay signal based on the loudness signal to obtain a loudness output signal Vo2.
It should be noted that, in the process of executing step S403, a multiplier may be used to multiply the loudness signal with the delay signal, so as to obtain the loudness output signal Vo2.
That is, the delay time of the delay time passing by the gain output signal Vo1 and the loudness output signal Vo2 must be kept consistent. Thereby ensuring that the high frequency signal and the low frequency signal are synchronized.
Based on this, when signal combining is performed subsequently, it can be ensured that the combined audio signal is a normal signal.
Step S104: and combining the gain output signal Vo1 and the loudness output signal Vo2 to obtain a sound source output signal Vout.
In performing step S104, the gain output signal Vo1 and the loudness output signal Vo2 may be combined based on an adder. Other devices capable of signal superposition may be used to achieve the above signal combination.
According to the embodiment of the invention, the low-frequency signal Vi1 and the high-frequency signal Vi2 are obtained by filtering the sound source input signal Vin, wherein the gain and the delay of the low-frequency signal Vi1 can be carried out to obtain the gain output signal Vo1, the loudness output signal Vo2 can be obtained by dynamic range control, delay and loudness control on the high-frequency signal Vi2, the obtained gain output signal Vo1 and the loudness output signal Vo2 are combined, and finally the signal Vout is output. The processing method disclosed by the embodiment of the invention can effectively improve the overall volume of the loudspeaker, so that the signal processing method disclosed by the embodiment of the invention can limit the displacement of the loudspeaker diaphragm and improve the overall volume.
Example two
Corresponding to the signal processing method provided in the embodiment of the present application, the embodiment of the present application further provides a corresponding signal processing device. As shown in fig. 5, a signal processing apparatus according to a second embodiment of the present application includes:
a filtering module 501, a high frequency signal processing module 502, a low frequency signal processing module 503 and a signal synthesizing module 504.
The filtering module 501 is configured to perform filtering processing on an audio input signal Vin to obtain a low-frequency signal Vi1 and a high-frequency signal Vi2, transmit the low-frequency signal Vi1 to the low-frequency signal processing module, and transmit the high-frequency signal Vi2 to the high-frequency signal processing module 504.
The low-frequency signal processing module 503 is configured to determine a gain signal based on the low-frequency signal Vi1, delay the gain signal according to a preset time, obtain a gain output signal Vo1, and use the gain output signal Vo1 as an input of the signal synthesizing module.
The high-frequency signal processing module 502 is configured to control the high-frequency signal Vi2 based on a dynamic range control algorithm, determine a loudness signal, delay the loudness signal according to a preset time, obtain a loudness output signal Vo2, and use the loudness output signal Vo2 as another input of the signal synthesizer;
the signal synthesizer 504 is configured to combine the gain output signal Vo1 and the loudness output signal Vo2 to obtain a sound source output signal Vout.
Referring to fig. 5, as shown in fig. 6, another signal processing apparatus according to an embodiment of the present invention is disclosed, where optionally, the signal synthesis module 504 includes an adder.
Optionally, the high-frequency signal processing module includes: a first multiplier 505, a dynamic range control module 506 and a first delay 507 in parallel.
The dynamic range control module 506 is configured to control the dynamic range of the high-frequency signal Vi2 according to a dynamic range control algorithm, determine a loudness signal that meets a preset loudness requirement, and use the loudness signal as an input of the multiplier 505, where the corresponding relationship between the dynamic range and the loudness is that the smaller the dynamic range is, and the larger the loudness is.
The first delayer 507 is configured to delay the input signal according to a preset time, obtain a delayed signal, and take the delayed signal as another input of the multiplier 505.
And the first multiplier 505 is configured to process the delay signal based on the loudness signal to obtain a loudness output signal.
Optionally, the low frequency signal processing module includes: a second multiplier 511, a first branch constituted by the second delayer 508, and a second branch constituted by the speaker model 509 and the gain calculation module 510 in series.
The speaker model 509 is configured to predict a diaphragm displacement M of an input signal of a sound source, and transmit the diaphragm displacement M to the gain calculation module.
The gain calculation module 510 is configured to determine whether the audio input signal needs gain according to the diaphragm displacement M, obtain a corresponding gain signal according to the determination result, and use the gain signal as an input of the second multiplier 511.
The second delayer 508 is configured to delay the input signal according to a preset time, obtain a delayed signal, and take the delayed signal as another input of the second multiplier 511.
The second multiplier 511 is configured to process the delay signal based on the gain signal to obtain a gain output signal.
Optionally, the filtering module 501 includes: a high pass filter 512 and a low pass filter 513 in parallel.
The high-pass filter 512 is configured to perform high-pass filtering on the audio input signal Vin to obtain a high-frequency signal Vi2, and take the high-frequency signal Vi2 as an input of the high-frequency signal processing module 502.
The low-frequency filter 513 is configured to perform low-pass filtering on the audio input signal Vin to obtain a low-frequency signal Vi1, and take the low-frequency signal Vi1 as an input of the low-frequency signal processing module 503.
It should be noted that, the high-pass filter set in the embodiment of the present invention can allow signals with signal frequency greater than 2KHz to pass, and the low-pass filter can only allow signals with signal frequency less than 2KHz to pass. The processing mode is that the audio signal passes through the high-pass filter and the low-pass filter simultaneously, and no time sequence exists.
According to the embodiment of the invention, the low-frequency signal Vi1 and the high-frequency signal Vi2 are obtained by filtering the sound source input signal Vin, wherein the gain and the delay of the low-frequency signal Vi1 can be carried out to obtain the gain output signal Vo1, the loudness output signal Vo2 can be obtained by dynamic range control, delay and loudness control on the high-frequency signal Vi2, the obtained gain output signal Vo1 and the loudness output signal Vo2 are combined, and finally the signal Vout is output. The processing method disclosed by the invention effectively improves the overall volume of the loudspeaker, so that the signal processing method disclosed by the invention can limit the displacement of the loudspeaker diaphragm and improve the overall volume.
Example III
Corresponding to the signal processing method and device provided by the embodiment of the application, the embodiment of the application also provides a corresponding loudspeaker. Fig. 7 shows a speaker according to a third embodiment of the present application, the speaker includes:
a signal processing device 701, a digital-to-analog converter 702, an audio power amplifier module 703 and a loudspeaker 704.
The signal processing device 701 is a signal processing device provided in the second embodiment, and is configured to perform filtering processing on an audio input signal Vin to obtain a low-frequency signal Vi1 and a high-frequency signal Vi2, transmit the low-frequency signal Vi1 to a low-frequency signal processing module, transmit the high-frequency signal Vi2 to the high-frequency signal processing module, determine a gain signal based on the low-frequency signal Vi1 by the low-frequency signal processing module, delay the gain signal according to a preset time, obtain a gain output signal Vo1, and input the gain output signal Vo1 from one end of the signal synthesis module; the high-frequency signal processing module controls the high-frequency signal Vi2 based on a dynamic range control algorithm, determines a loudness signal, delays the loudness signal according to preset time to obtain a loudness output signal Vo2, and inputs the loudness output signal Vo2 from the other end of the signal synthesizer. The signal synthesizer combines the gain output signal Vo1 and the loudness output signal Vo2 to obtain the sound source output signal Vout.
The input end of the digital-analog converter 702 is connected to the output end of the signal processing device 701, and the output end is connected to the input end of the audio power amplifier module 503.
The output end of the audio power amplifier module 703 is connected with the input end of the loudspeaker 704.
According to the embodiment of the invention, the audio input signal Vin is filtered through the signal processing device to obtain the low-frequency signal Vi1 and the high-frequency signal Vi2, wherein the low-frequency signal Vi1 is subjected to gain and delay to obtain the gain output signal Vo1, the high-frequency signal Vi2 is subjected to dynamic range control, delay and loudness control to obtain the loudness output signal Vo2, the gain output signal Vo1 and the loudness output signal Vo2 are finally combined to output the signal Vout, the output signal Vout is transmitted to the audio power amplifier after passing through the digital-analog converter, and finally sound is output through the loudspeaker. The processing method disclosed by the invention effectively improves the overall volume of the loudspeaker, so that the signal processing method disclosed by the invention can limit the displacement of the loudspeaker diaphragm and improve the overall volume.
In this specification, each embodiment is described in a progressive manner, and identical and similar parts of each embodiment are all referred to each other, and each embodiment mainly describes differences from other embodiments. In particular, for a system or system embodiment, since it is substantially similar to a method embodiment, the description is relatively simple, with reference to the description of the method embodiment being made in part. The systems and system embodiments described above are merely illustrative, wherein the elements illustrated as separate elements may or may not be physically separate, and the elements shown as elements may or may not be physical elements, may be located in one place, or may be distributed over a plurality of network elements. Some or all of the modules may be selected according to actual needs to achieve the purpose of the solution of this embodiment. Those of ordinary skill in the art will understand and implement the present invention without undue burden.
Those of skill would further appreciate that the various illustrative elements and algorithm steps described in connection with the embodiments disclosed herein may be implemented as electronic hardware, computer software, or combinations of both, and that the various illustrative elements and steps are described above generally in terms of functionality in order to clearly illustrate the interchangeability of hardware and software. Whether such functionality is implemented as hardware or software depends upon the particular application and design constraints imposed on the solution. Skilled artisans may implement the described functionality in varying ways for each particular application, but such implementation decisions should not be interpreted as causing a departure from the scope of the present invention.
The previous description of the disclosed embodiments is provided to enable any person skilled in the art to make or use the present invention. Various modifications to these embodiments will be readily apparent to those skilled in the art, and the generic principles defined herein may be applied to other embodiments without departing from the spirit or scope of the invention. Thus, the present invention is not intended to be limited to the embodiments shown herein but is to be accorded the widest scope consistent with the principles and novel features disclosed herein.
Claims (8)
1. A method of signal processing, the method comprising:
filtering the sound source input signal Vin to obtain a low-frequency signal Vi1 and a high-frequency signal Vi2;
determining a gain signal based on the low-frequency signal Vi1, and delaying the gain signal according to preset time to obtain a gain output signal Vo1;
controlling the high-frequency signal Vi2 based on a dynamic range control algorithm, determining a loudness signal, and delaying the loudness signal according to preset time to obtain a loudness output signal Vo2;
combining the gain output signal Vo1 and the loudness output signal Vo2 to obtain a sound source output signal Vout;
the determining a gain signal based on the low frequency signal Vi1 and delaying the gain signal according to a preset time to obtain a gain output signal Vo1 includes: obtaining diaphragm displacement M of a predicted sound source input signal according to a loudspeaker model;
judging whether the sound source input signal needs gain according to the diaphragm displacement M, and obtaining a corresponding gain signal according to a judging result;
delaying the sound source input signal according to preset time to obtain a delay signal;
and processing the delay signal based on the gain signal to obtain a gain output signal Vo1.
2. A method according to claim 1, wherein the controlling the high frequency signal Vi2 based on the dynamic range control algorithm, determining a loudness signal, and delaying the loudness signal according to a preset time to obtain a loudness output signal Vo2, comprises: controlling the dynamic range of the high-frequency signal Vi2 according to a dynamic range control algorithm, and determining a loudness signal meeting the preset loudness requirement;
delaying the sound source input signal according to preset time to obtain a delay signal;
and processing the delay signal based on the loudness control to obtain a loudness output signal Vo2.
3. Method according to any of claims 1-2, characterized in that the filtering of the audio source input signal Vin to obtain a low frequency signal Vi1 and a high frequency signal Vi2 comprises: performing high-pass filtering processing on the sound source input signal Vin to obtain a high-frequency signal Vi2;
and performing low-pass filtering processing on the sound source input signal Vin to obtain a low-frequency signal Vi1.
4. A signal processing apparatus, characterized in that the signal processing apparatus comprises: the device comprises a filtering module, a high-frequency signal processing module, a low-frequency signal processing module and a signal synthesizing module;
the filtering module is used for filtering the sound source input signal Vin to obtain a low-frequency signal Vi1 and a high-frequency signal Vi2, transmitting the low-frequency signal Vi1 to the low-frequency signal processing module and transmitting the high-frequency signal Vi2 to the high-frequency signal processing module;
the low-frequency signal processing module is used for determining a gain signal based on the low-frequency signal Vi1, delaying the gain signal according to preset time to obtain a gain output signal Vo1, and inputting the gain output signal Vo1 as one end of the signal synthesizing module;
the high-frequency signal processing module is used for controlling the high-frequency signal Vi2 based on a dynamic range control algorithm, determining a loudness signal, delaying the loudness signal according to preset time to obtain a loudness output signal Vo2, and inputting the loudness output signal Vo2 as the other end of the signal synthesis module;
the signal synthesis module is configured to combine the gain output signal Vo1 and the loudness output signal Vo2 to obtain a sound source output signal Vout;
the low frequency signal processing module includes: the second multiplier is a first branch composed of a second delayer, and a second branch composed of a loudspeaker model and a gain calculation module which are connected in series;
the loudspeaker model is used for predicting the vibration film displacement M of the sound source input signal and transmitting the vibration film displacement M to the gain calculation module;
the gain calculation module is configured to determine whether the sound source input signal Vin needs gain according to the diaphragm displacement M, obtain a corresponding gain signal according to a determination result, and use the gain signal as an input of the second multiplier;
the second delayer is configured to delay the input signal Vi1 according to a preset time, obtain a delay signal Vi1, and use the delay signal as another input of the second multiplier;
the second multiplier is configured to process the delay signal based on the gain signal to obtain a gain output signal Vo1.
5. The apparatus of claim 4, wherein the high frequency signal processing module comprises: the first multiplier, the dynamic range control module and the first delay device are connected in parallel;
the dynamic range control module is used for controlling the dynamic range of the high-frequency signal Vi2 according to a dynamic range control algorithm, determining a loudness signal Vo2 meeting the preset loudness requirement, and taking the loudness signal Vo2 as one input of the multiplier;
the first delayer is used for delaying the input signal according to preset time to obtain a delay signal, and taking the delay signal as the other input of the multiplier;
and the first multiplier is used for processing the delay signal based on the loudness signal Vo2 to obtain a loudness output signal Vo2.
6. The apparatus of any one of claims 4-5, wherein the filtering module comprises: a high pass filter and a low pass filter connected in parallel;
the high-pass filter is used for performing high-pass filtering processing on the sound source input signal Vin to obtain a high-frequency signal Vi2, and taking the high-frequency signal Vi2 as the input of the high-frequency signal processing module;
the low-pass filter is configured to perform low-pass filtering processing on the audio input signal Vin to obtain a low-frequency signal Vi1, and take the low-frequency signal Vi1 as an input of the low-frequency signal processing module.
7. The apparatus of any of claims 4-5, wherein the signal synthesis module comprises an adder.
8. A loudspeaker, comprising: the signal processing device of any one of claims 4-7, a digital-to-analog converter, an audio power amplifier module, and a speaker;
the input end of the digital-analog converter is connected with the output end of the signal processing device, and the output end of the digital-analog converter is connected with the input end of the audio power amplifier module;
the output end of the audio power amplifier module is connected with the input end of the loudspeaker.
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| CN115567831A (en) * | 2021-06-30 | 2023-01-03 | 华为技术有限公司 | Method and device for improving tone quality of loudspeaker |
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