Method for converting audio signal characteristics and associated device

Abstract

A method and related apparatus for converting multiple characteristics of an audio signal are disclosed. The present invention relates to a method and related apparatus for combined conversion of multiple characteristics of an audio signal. The variation allows for assignment of the signal according to a configuration scheme selected by a control unit. The method and apparatus according to the present invention are particularly used in the field of loudspeakers.

Description

Technical Field

[0001] The present invention relates to a method for converting in a combined manner multiple characteristics of an audio signal intended for a loudspeaker, and an associated device. The device includes a processor and an amplifier for all or part of the frequency bands. The processor is connected to a control module that allows selection of a signal characteristic conversion mode. Background Art

[0002] "Loudspeaker" generally refers to all types of electroacoustic transducers and mechanical acoustic transducers.

[0003] According to the invention with the patent number US669,749,2, it discloses an acoustic loudspeaker system with digital signal processing function.

[0004] This system compares the output signal with the input signal through a sensor. The comparison result is used to make the output signal consistent with the input signal through correction.

[0005] This equalization device allows correction of the signal gain (dB) within certain frequency bands, and the correction coefficient is adapted to each bandwidth of the loudspeaker to be corrected.

[0006] The main drawback of this device is that it only acts on the gain parameter (dB). Although this correction can achieve linearity of the gain / frequency ratio, it is still unsatisfactory in all other parameters characterizing the signal complexity such as phase and time. In fact, the non-linearity of phase and time will prevent faithful reproduction of the original sound.

[0007] According to the invention with the patent number JP2571091, it discloses a frequency characteristic correction device for a loudspeaker. According to the invention with the patent number JP2530474, it discloses a method related to the device. The device and method allow correction of the gain (dB) and phase of the signal across the entire spectrum. A digital adaptive system intervenes at each frequency to linearize the amplitude / frequency curve and the phase / frequency curve. The device can continuously correct the signal with the help of a sensor.

[0008] The drawback of such continuous correction is the delay in the processing process, so it cannot act on signals with a reproduction time shorter than the processing time.

[0009] In addition, stray signals such as noise in the room can interfere with the processing.

[0010] According to the invention with the patent number CA2098319, it discloses an analog signal processing device for correcting inaccuracies in harmonics and phase caused by energy conversion, recording, and on-site playback of audio signals.

[0011] Among them, by automatically and continuously performing calibration, the realism of the reproduced audio signal is restored.

[0012] Permanent and continuous calibration makes the type of music being listened to unable to be adjusted, and different processing is required to achieve adjustment.

[0013] According to the invention with the publication number US2015073574, it discloses a method that allows access to a content stream to be distributed to a playback device and then identifies the content that allows a predetermined configuration scheme to be passed to it.

[0014] Depending on the identified configuration scheme, the method allows the correction of equalization parameters related to the playback of the content stream.

[0015] This scheme can adjust the equalization situation with respect to the available information identified during the playback process, or related to the user configuration scheme, or related to the audio support set by the user setting scheme.

[0016] The main drawback of this method is that it only provides equalization correction, that is, the correction of gain (expressed in dB) according to frequency. This correction is still unsatisfactory in terms of all other parameters characterizing the signal complexity, such as phase and time. Summary of the Invention

[0017] In view of this, the object of the present invention is to make up for the above-mentioned various drawbacks. More specifically, the object of the present invention is to provide a method and an associated device that allow the correction of all characteristics of a complex signal, such as:

[0018] Gain;

[0019] Phase;

[0020] Time;

[0021] Distortion;

[0022] Bandwidth;

[0023] Speaker bandwidth distribution;

[0024] Dynamic compression / expansion;

[0025] Directivity;

[0026] Sampling;

[0027] Absolute phase corresponding to the electrical polarity of the speaker group under the impulse response;

[0028] Offset of the reference point when all frequencies are in phase.

[0029] By the combined application of such corrections, the sound can be accurately, instantaneously imparted, compensated, or improved according to a typical configuration scheme.

[0030] "Imbuing" generally refers to endowing an audio signal with specific characteristics.

[0031] This method can convert various audio signal characteristics in a combined manner and decompose them into a series of operations that can be performed in one or more stages.

[0032] The first operation is to create a correction scheme designed to linearize the output signal, taking into account the inherent defects of the speaker components and structure. "Speaker" refers to a speaker group composed of one or more speakers installed in a closed or open structure.

[0033] The subsequent second operation is to implement corrections related to the overall signal characteristics according to a predetermined configuration scheme.

[0034] These two signal conversion operations can be performed in the same step so that all selected conversion items can be directly implemented.

[0035] These correction items can also be carried out in multiple steps so that the correction operations for neutralizing the signal are separated from the correction operations for adding imbuing, compensation, or improvement effects. In this way, each operation will be easier to control. On the other hand, since the application basis of the correction scheme is the neutral signal, the standardization of the correction scheme can be achieved.

[0036] The present invention relates to a method for converting various characteristics of an audio signal intended for a speaker in a combined manner, the method comprising the following operations:

[0037] A first correction operation for measuring the output signal of the speaker to determine the defects to be corrected according to a reference template, and then generating a correction scheme, which is then used to linearize all characteristics, for example, to achieve gain, phase, time equalization, and minimization of the distortion level. In this way, the correction performed will vary depending on the speaker used;

[0038] A second operation, including correcting the previously obtained neutral signal to adapt it to a given configuration scheme; this correction can be completed according to one or more criteria, such as: gain; phase; time; distortion; bandwidth; speaker bandwidth distribution; dynamic range compression / expansion; directivity; sampling; reference phase corresponding to the polarity of the speaker group under the impulse response; and the offset of the reference point when all frequencies are in phase.

[0039] According to an advantageous but non-mandatory aspect of the present invention, such a method may include one of the following features, or multiple features combined in any technically permissible manner:

[0040] The control module can be manually operated by the user;

[0041] The control module can perform automatic adjustment by selecting a typical configuration scheme according to the music style information included in the music track;

[0042] The control module can perform automatic adaptation according to the information for identifying signals and the information for identifying typical configuration schemes included in the remote service;

[0043] The control module can perform automatic adaptation according to the user preferences identified by the above device;

[0044] The control module performs automatic adaptation according to the information received by a sensor that is provided in the device or at a remote location and measures weather conditions such as atmospheric temperature, atmospheric pressure, or humidity.

[0045] The present invention also relates to an associated device for converting various characteristics of an audio signal of a loudspeaker in a combined manner. The device includes a signal conversion module for all or part of the frequency bands. The conversion module is connected to a control module that selects a signal characteristic conversion mode according to a predetermined configuration scheme.

[0046] According to an advantageous but non-compulsory aspect of the present invention, such a device may include one of the following features, or a combination of multiple features in any technically permissible manner:

[0047] The signal conversion can be implemented by a processor according to a digital method;

[0048] The signal conversion can be implemented by electrical and / or electronic components according to an analog method;

[0049] The signal conversion can be implemented according to one or more mechanical methods using a tuning structure, an acoustic lens, and / or the geometric characteristics of the device. BRIEF DESCRIPTION OF THE DRAWINGS

[0050] According to the following detailed description, other features and advantages of the present invention will become readily apparent. The following description is intended to be understood with reference to the accompanying drawings, in which:

[0051] Figure 1 is a schematic diagram of the device of the present invention;

[0052] Figure 2 shows the steps of a general signal conversion method;

[0053] Figure 3 shows the conversion of the frequency characteristics of an audio signal by Figure 2 method;

[0054] Figure 4 shows the conversion of the phase characteristics of an audio signal by Figure 2 method;

[0055] Figure 5 Shown is the conversion of the temporal characteristics of an audio signal by Figure 2 a method;

[0056] Figure 6 Shown is the conversion of the bandwidth characteristics of an audio signal by Figure 2 a method;

[0057] Figure 7 Shown is the conversion of the compression / expansion characteristics of an audio signal by Figure 2 a method;

[0058] Figure 8 Shown is the conversion of the distortion characteristics of an audio signal by Figure 2 a method;

[0059] Figure 9 Shown is the conversion of the directivity characteristics of an audio signal by Figure 2 a method;

[0060] Figure 10 Shown is the conversion of the sampling characteristics of an audio signal by Figure 2 a method;

[0061] Figure 11 Shown is the conversion of the absolute phase characteristics of an audio signal by Figure 2 a method;

[0062] Figure 12 Shown is the conversion of the reference point characteristics of all frequencies of an audio signal by Figure 2 a method;

[0063] Figure 13 Shown is the conversion of an audio signal involving cut-off frequency correction by Figure 2 a method. Detailed implementation manner

[0064] Referring to Figure 1 , the device of the present invention includes a processor 1 for at least one frequency band. The processor is, for example, a digital or analog signal processor 1 (in the form of, for example, discrete filters), and receives an audio signal in a wired or wireless manner. The audio signal can be either an analog signal or a digital signal. In Figure 1 , this obtained audio signal is denoted as IN.

[0065] The signal processor 1 can perform processing in an analog manner using electrical or electronic components, or in a digital manner using a processor such as a digital signal processor (DSP) or a microcontroller module. The signal can be power-amplified in an analog or digital manner by the amplifier 2. In the case of conversion from the analog domain to the digital domain, a converter (not shown in the figure) must be added to convert the signal from an analog signal to a digital signal.

[0066] This electrical signal is finally converted into an acoustic signal by an electroacoustic transducer (also called a mechanical acoustic transducer) such as the speaker 3.

[0067] According to each embodiment such as the embodiment shown as Figure 1 This device may include a signal processing chain, and this signal processing chain may include such a processor 1, such an amplifier 2, and such a transducer 3 for each frequency band B1, Bn.

[0068] Therefore, it can be understood that in this case, for each frequency band B1, Bn, this device includes a dedicated processor 1, amplifier 2, and transducer 3.

[0069] Alternatively, this device includes a common processor 1, amplifier 2, and transducer 3 for all frequency bands.

[0070] This device is equipped with a control module 4, also called a mode decoder, for selecting a signal change and automatically or manually applying it to this device or disabling it. Such a selection can be completed by the user through a selection module 7 including a human-machine interface, for example.

[0071] In the automatic mode, this device can either receive a configuration scheme from a remote service 5, such as Gracenote (registered trademark), Shazam (registered trademark), or any equivalent service (refer to the patent with the publication number US2015073574), or select a configuration scheme by using an identification system of an internal database, or through artificial intelligence.

[0072] Optionally, this device may be equipped with a mechanical or acoustic system 6 for correcting the physical characteristics of this device. This correction system 6 can be achieved, for example, by correcting the size of the acoustic load, setting an acoustic lens composed of one or more deflectors, correcting the characteristics of the resonator, or any equivalent measure.

[0073] The system 6 generally includes a mechanical acoustic processor 6-1 and a mechanical acoustic driver 6-2.

[0074] Generally speaking, the device of the present invention is capable of performing a combined conversion on a plurality of audio signal characteristics selected in a non-limiting manner from the following characteristics:

[0075] - Gain;

[0076] - Phase;

[0077] - Time;

[0078] - Distortion;

[0079] - Bandwidth;

[0080] - Bandwidth distribution of each speaker;

[0081] - Dynamic compression / expansion;

[0082] - Directivity;

[0083] - Sampling;

[0084] - Absolute phase corresponding to the electrical polarity (connection polarity) of the speaker group under impulse response;

[0085] - Offset of the reference point when all frequencies are in phase.

[0086] Combinations of such changes in various audio signal characteristics can accurately and instantaneously assign, compensate for, or improve the corresponding sounds according to typical configuration schemes. "Assignment" means endowing the audio signal with specific characteristics.

[0087] Figure 2 The flowchart shows a general signal conversion method including a calibration operation and another correction operation according to an embodiment of the present invention.

[0088] The execution of the steps in this conversion method is controlled, for example, by the control module 4 in the device of the present invention.

[0089] The starting step of this method is step 100, in which the output signal of the speaker is measured. This measurement can be carried out in the laboratory during the design of this device with the help of a system composed of a generator, a microphone, and a signal processing system connected to a computer, and the computer executes information acquisition and processing software.

[0090] Subsequently, in step 102, the defects to be corrected are determined by analyzing the differences between the input signal and the reference template. The reference template is an ideal curve of relevant characteristics such as gain, phase, time, and distortion.

[0091] After that, in step 104, a calibration scheme is derived based on the above analysis and the selected criteria. Depending on the selected type of processing, it can include the application of digital processing algorithms, analog processing plans composed of a group of electrical and / or electronic components, or algorithms for controlling the mechanical system 6.

[0092] Thereafter, in step 106, the system linearizes all the characteristics of the signal by applying the correction scheme to reproduce the neutrality it originally had. Depending on the type of processing selected, this scheme can be directly applied by processor 1 in the case of digital processing, applied through active or passive filtering in analog processing, or applied by mechanical system 6 capable of changing the geometric characteristics of the present device.

[0093] After the signal is linearized, in step 108, the assignment of characteristics is performed by applying a correction scheme according to the selected configuration scheme. Such schemes are created in advance through feedback methods that respectively depend on the various configuration schemes sought (such as music type, recording type, reproduction or atmosphere type). Such schemes are selected, for example, after the configuration scheme is obtained in advance (step 110), according to the configuration scheme selected by the user in manual mode or by control module 4 in automatic mode. In automatic mode, the present device can receive the configuration scheme from remote service 5 or an internal database (step 112).

[0094] Subsequently, in step 114, one or more of the amplifiers 2 perform power amplification of the above signal in analog or digital mode.

[0095] Finally, in step 116, the electrical signal is converted into an acoustic signal by speaker 3 or any equivalent transducer.

[0096] Optionally, control module 4 automatically adjusts according to the information received by sensors provided in the present device or at a remote location and measuring weather conditions such as atmospheric temperature, atmospheric pressure or humidity.

[0097] Figure 3 The curve shown is the conversion of the amplitude (vertical axis) / frequency (horizontal axis) curve of the measured exemplary audio signal within different conversion stages.

[0098] Figure 3 Subfigure (a) of shows an example of the signal measured in step 100 above. For example, this signal is not ideal due to the inherent characteristics of the components of the present device. In the prior art, all speakers cause distortion of the signals they process.

[0099] Figure 3 Subfigure (b) of shows the above curve corrected, for example, in step 106. This curve is obtained with the goal of leveling all amplitudes as much as possible as the frequency changes. In the case of analog processing, this correction operation is implemented by functional devices such as filters, for example, by a resonant tank circuit. In the case of digital processing, this correction operation is implemented by a digital signal processor such as a DSP. For each frequency processed, the digital signal processor corrects the gain of the signal. In the case of mechanical processing, tuning structures such as cavities, resonators, baffles and / or mufflers can be used.

[0100] Figure 3 Sub - figure (c) shows an example of the curve after correction in step 108. This amplitude - corrected graph is obtained according to feedback in the case of sound recording or reproduction. In the case of analog processing, this correction operation is implemented by functional devices such as filters, for example, by a resonant tank circuit. In the case of digital processing, this correction operation is implemented by a digital signal processor such as a DSP. For each frequency being processed, the digital signal processor corrects the gain of the signal. In the case of mechanical processing, tuning structures such as cavities, resonators, baffles, and / or mufflers can be used.

[0101] Figure 4 The curve shown is Figure 3 the conversion of the phase (vertical axis) / frequency (horizontal axis) curve of the signal in the above - mentioned different conversion steps.

[0102] Figure 4 Sub - figure (a) shows the signal measured in step 100. Similarly, this signal is not ideal due to the inherent characteristics of the components of this device. In the prior art, all speakers cause distortion of the signals they process.

[0103] Figure 4 Sub - figure (b) shows the above - mentioned curve after correction in step 106. This curve is obtained with the goal of leveling all phases as much as possible as the frequency changes. In the case of analog processing, this correction operation is implemented by functional devices such as filters, for example, by a phase circuit. In the case of digital processing, this correction operation is implemented by a digital signal processor such as a DSP. For each frequency being processed, the digital signal processor corrects the phase of the signal. In the case of mechanical processing, tuning structures such as cavities, resonators, baffles, and / or mufflers can be used.

[0104] Figure 4 Sub - figure (c) shows an example of the curve after correction in step 108. This phase - corrected graph is obtained by approaching the phase change of the recording studio speaker or the reproduction speaker. In the case of analog processing, this correction operation is implemented by functional devices such as filters, for example, by a phase circuit. In the case of digital processing, this correction operation is implemented by a digital signal processor such as a DSP. For each frequency being processed, the digital signal processor corrects the phase of the signal. In the case of mechanical processing, tuning structures such as cavities, resonators, deflectors, and / or mufflers can be used.

[0105] Figure 5 The curve shown is the time (vertical axis) / frequency (horizontal axis) curve conversion of the measured exemplary audio signal within different conversion steps.

[0106] Figure 5 Sub - figure (a) shows an example of the signal measured in the above - mentioned step 100. For example, this signal is not ideal due to the inherent characteristics of the components of this device. In the prior art, all speakers cause distortion of the signals they process.

[0107] Figure 5 Sub - figure (b) shows the above - mentioned curve corrected, for example, in step 106. This curve is obtained with the goal of leveling all times as evenly as possible as the frequency changes. In the case of analog processing, this correction operation is implemented by functional devices such as filters, for example, by a phase circuit that corrects over time. In the case of digital processing, this correction operation is implemented by a digital signal processor such as a DSP. For each frequency processed, this digital signal processor corrects the time of the signal. In the case of mechanical processing, the speaker is physically displaced in space, and tuning structures such as cavities, resonators, baffles, and / or mufflers can also be used.

[0108] Figure 5 Sub - figure (c) shows an example of the curve after being corrected in step 108. This phase - correction graph is obtained by approaching the time variation of a studio speaker or a reproduction speaker. In the case of analog processing, this correction operation is implemented by functional devices such as filters, for example, by a phase circuit. In the case of digital processing, this correction operation is implemented by a digital signal processor such as a DSP. For each frequency processed, this digital signal processor corrects the time of the signal.

[0109] More precisely, the purpose of the above - mentioned processing is to perform time correction for each frequency band in the frequency decomposition (or analysis) of the signal.

[0110] In the case of mechanical processing, the speaker is physically displaced in space, and tuning structures such as cavities, resonators, baffles, and / or mufflers can also be used.

[0111] Figure 6 shows the frequency - response signal curve of an exemplary audio signal, used to illustrate the bandwidth - curve conversion performed by the Figure 2 method. The solid line represents the first response signal corresponding to the frequency response that a transducer usually exhibits according to its inherent performance.

[0112] In contrast, the dashed lines represent two corrected signals corresponding to shortening or lengthening the response curve respectively.

[0113] On the one hand, the above curve can be shortened (narrowed) at low and high frequencies to protect the speaker and limit mechanical distortion that has an adverse effect on the remaining frequency spectrum. In the case of analog processing, this bandwidth shortening effect can be achieved through functional devices such as filters, for example, by means of high-pass and / or low-pass circuits. In the case of digital processing, this correction operation is achieved by running high-pass and / or low-pass filtering algorithms on a digital signal processor such as a DSP. In the case of mechanical processing, tuned structures such as cavities, resonators, acoustic short plates, and / or mufflers can be used.

[0114] On the other hand, the above curve can be lengthened (broadened) as much as possible to better achieve the restoration of the sound signal. In the case of analog processing, this bandwidth expansion effect can be achieved through functional devices such as resonant circuits. In the case of digital processing, this correction operation is achieved by running filtering algorithms capable of achieving gain on a digital signal processor such as a DSP. In the case of mechanical processing, tuned structures such as cavities, resonators, acoustic short plates, and / or horns can be used.

[0115] Figure 7 For Figure 2 Schematic diagram of the conversion curve for converting the compression or expansion characteristics of the exemplary signal by the method shown. These curves are represented as the variation curves of the output signal OUT (vertical axis) with respect to the input signal IN (horizontal axis).

[0116] Figure 7 Subfigure (a) shows the compression curve obtained after compressing the measured signal. In the compression mode, as the input signal gradually increases, the amplification factor of the circuit under consideration gradually decreases until it becomes negative. Thus, a very significant level control effect can be achieved. In the case of analog processing, signal compression can be achieved through functional devices such as compression circuits, for example, by an amplifier whose gain can vary with the input level. In the case of digital processing, signal compression can be achieved by running a compression algorithm on a digital signal processor such as a DSP.

[0117] Figure 7 Subfigure (b) shows the expansion curve obtained after expanding the measured signal. In the expansion mode, the amplification factor of the circuit under consideration increases with the increase of the input signal. Thus, it has the effect of restoring the dynamics of the compressed signal, thereby improving its spirituality. In the case of analog processing, signal expansion can be achieved through functional devices such as expansion circuits, for example, by an amplifier whose gain can vary with the input level. In the case of digital processing, signal expansion can be achieved by running an expansion algorithm on a digital signal processor such as a DSP.

[0118] Figure 8 Shown as Figure 2Method for correcting a signal conversion curve obtained when measuring distortion characteristics of an exemplary audio signal.

[0119] Figure 8 Sub - figure (a) shows a spectrogram composed of the fundamental frequency F and its harmonics Hn that cause a high distortion rate. A high distortion rate means that unwanted signals not present in the original signal are added. The main causes of a high distortion rate are electrical and mechanical defects in the reproduction system, or are mainly caused by the non - linearity of the system's phase and time. In addition, by increasing the distortion rate of the signal, defects not present in the original signal can be simulated, i.e., coloring the sound. "Coloring" generally refers to endowing an audio signal with specific characteristics. By causing distortion in a controllable manner, for example, the harmonic distortion characteristics of a high - performance speaker can be mimicked. In the case of analog processing, by adding multiple frequencies to a selected fundamental frequency, an increase in the distortion degree is achieved. In the case of digital processing, by running an algorithm for generating harmonic frequencies on a digital signal processor such as a DSP, an increase in the distortion degree is achieved.

[0120] Figure 8 Sub - figure (b) shows a spectrogram composed of the fundamental frequency F and its harmonics with a reduced distortion rate after conversion. A low distortion rate means that the reproduced signal is closer to the original signal. In the case of analog processing, by means of a filtering function or phase and time correction, unwanted frequencies are suppressed, thereby obtaining a reduction in the distortion degree. In the case of digital processing, by executing filtering and / or phase and time correction algorithms on a digital signal processor such as a DSP, a reduction in the distortion degree is obtained.

[0121] Figure 9 Shown are different orientations of the sound emitted from the speaker HP according to different directivity characteristics.

[0122] Figure 9 Sub - figure (a) shows an open horizontal direction pattern, highlighting the proportion of reflected sounds that interfere with the direct sound due to the scattering of sound on the wall M.

[0123] Figure 9 Sub - figures (b) and (c) show more closed direction patterns that limit the reflection of sound on the wall M. The listener A will hear more direct sounds compared to the reflected sounds. This result is achieved by combining mechanical / acoustic measures and electrical measures, such as adding speakers and waveguides, and / or controlling the time and phase differences between the two.

[0124] Figure 10 The curves S1 and S2 shown are curves of the sampling signal amplitude (vertical axis) varying with time (horizontal axis). The reference curve S represents the corresponding analog signal before sampling.

[0125] Figure 10The coarsely sampled curve of curve S1 in terms of time and quantization as shown in sub - figure (a). For example, it is the CD standard characterized by a 16 - bit format with a sampling frequency of 44.1 kHz.

[0126] Figure 10 The more finely sampled curve of curve S1 in terms of time and quantization as shown in sub - figure (b). This conversion is achieved by increasing the number of bits (e.g., from 16 bits to 24 bits) and increasing the number of samples per unit time (e.g., increasing the sampling frequency from 44.1 kHz to 192 kHz). This conversion can reduce the distortion rate by interpolating the signal and thus reducing the increment step. In this way, the listening comfort is improved. This conversion is implemented digitally by an asynchronous sample rate converter (more well - known by its acronym "ASRC").

[0127] In Figure 11 In the method shown, the illustrated absolute phase position corresponds to the electrical polarity of the speaker group under the impulse response, thus realizing the correction of the perception of the depth of the sound scene.

[0128] Figure 11 The negative impulse response I - that makes people feel the sound is in the near - field (position P1) is shown in sub - figure (a).

[0129] Figure 11 The positive impulse response I + that makes people feel the scene depth (position P2) is larger is shown in sub - figure (b).

[0130] By reversing the connection polarity of the speaker group, the above two can be interchanged.

[0131] In Figure 12 In the method shown, the illustrated position is the reference phase position.

[0132] Figure 12 Multiple possible positions C1, C2, C3 of the reference phase are shown. Depending on the required relative position with devices such as the speaker HP, the line of the reference phase is at 0 degrees. For example, this position can be at a negative distance, and this distance is approximately the distance that makes people feel the scene depth is larger. In addition, it can also be at a positive distance, and this distance is approximately the distance that makes people feel the scene is in the near - field.

[0133] This conversion can be digitally implemented by a processor such as a DSP that recalculates the correct phase at the selected distance.

[0134] Figure 13 The different bandwidth distributions of each speaker are shown, and this bandwidth distribution corresponds to the cut - off frequency or the frequency offset.

[0135] Figure 13Sub - figure (a) shows a situation where, due to the crossover frequency FC1 shifting towards the bass (low - frequency) direction, the distortion rate of the device increases and the directivity weakens.

[0136] Figure 13 Sub - figure (b) shows a situation where, due to the bandwidth being evenly distributed (the cut - off frequency FC2 is basically in the center of the frequency band), a balance in the usage area is achieved between different speakers considering mechanical limitations, electrical limitations, power - handling capacity limitations, and / or directivity limitations.

[0137] Figure 13 Sub - figure (c) shows a situation where, due to the crossover frequency FC3 shifting towards the high - frequency direction of the audio frequency band, the speakers designed to receive such frequencies are protected by receiving less energy. In addition, this situation also increases the directivity of the device.

[0138] In all of the above three situations, the shift and slope of the crossover frequency are achieved by changing the filter type and parametrically in both analog and digital aspects.

[0139] In many embodiments, the control module automatically adjusts the selection of the typical configuration according to information related to the specific music style of the track. That is, the control module is configured to automatically identify the music style of the played signal. In this way, the control module can determine the type of music being played and automatically adjust its settings to fit the recording conditions and the type of the work being played. This description is particularly applicable to the case where the system includes two independent active multi - channel speakers (left / right).

[0140] The identification of music is achieved, for example, by sampling the signal; then, analyzing the signal in one or more possible ways, such as online services or applications like Shazam or Gracenote (registered trademark); and comparing the music sample with reference data stored in a remote database via an Internet connection or a local database. The determination of the music type can also be achieved through the information contained in the music file (e.g., ID3 tags in the MP3 format), or through other determination methods, such as determination algorithms based on one or more features of the music (rhythm, harmonic content, etc.).

[0141] The recognition method can be different, for example, depending on whether the recognition is performed within the receiving device (loudspeaker) or within the transmitting device. In the case of a wireless link, if the recognition is performed within the receiving device, synchronization must be carried out between the individual receiving devices to avoid inconsistencies in the set content between the receiving devices. The preferably used mode is the "master-slave" mode: the "master" device is responsible for determining the music type and the set content to be applied, and sharing the result with the "slave" devices; each "slave" device stores the requested setup program therein and applies it. In addition, the analysis can also be performed within the transmitting device. In this case, the state of the transmitting device is the "master device". After recognizing the music style, the control module selects a typical configuration corresponding to the recognized music style. The typical configuration can be a set of set contents or a "scheme" for one or more signal characteristics, and such set contents change the behavior of the loudspeaker in a combined manner. Therefore, one loudspeaker can have a similar behavior acoustically to another loudspeaker designed in a different way or for a different music type. The loudspeaker can have some (for example, four) basic set contents predefined by the loudspeaker manufacturer at the factory, and then the user can update them.

[0142] In practical applications, the above set contents can include some or all of the following elements: gain; phase; time; distortion; bandwidth; bandwidth distribution of each loudspeaker; dynamic compression; directivity; absolute phase; equalization.

[0143] For example, the typical configuration corresponding to the music style called current music can have the following set contents:

[0144] - Gain: Increase the gain of the signal in the high-frequency channel;

[0145] - Phase: Retain the phase rotation caused by each filter (without correcting it); shift the cut-off frequency of the filter between the bass and midrange signals so that the required energy must be maintained at the connection point by adjusting the phase curve;

[0146] - Time: Also retain (without correction) the inherent time steps of the acoustic load and the filter;

[0147] - Distortion: The mechanical distortion rate of the loudspeaker and the phase and time distortion can be controlled or limited by the selection of the filter, slope or type;

[0148] - Bandwidth: The high-pass filter filters out frequencies below 60 Hz;

[0149] - The bandwidth distribution of each loudspeaker is selected so that the bass signal overlaps with the midrange signal at their connection frequency; for example, for a connection frequency of 150 Hz, the cut-off frequency of the bass transducer is 200 Hz, and the starting frequency of the midrange transducer is 100 Hz;

[0150] - Compression: Limit the dynamic difference between the peak amplitude and the average amplitude;

[0151] - Directivity: Shift the cut-off frequency between the midrange and the treble upward by an octave;

[0152] - Absolute phase: Do not reverse the polarity of the speaker;

[0153] - Equalization:

[0154] For 42.5 Hz, +2.5 dB, Q factor = 3.4

[0155] For 200 Hz, -0.5 dB, Q factor = 2.2

[0156] For 3400 Hz, +1.5 dB, Q factor = 0.71

[0157] For 20000 Hz, +5.0 dB, Q factor = 0.50

[0158] In addition, there are other feasible examples.

[0159] For example, a typical configuration corresponding to a music style called acoustic music may have the following settings:

[0160] - Gain: The gain setting of the signal is selected so that there is no amplitude difference between the frequency bands;

[0161] - Phase: Eliminate the phase rotation caused by the load and each filter through correction (e.g., with DSP);

[0162] - Time: For each frequency band, adjust the signal processing delay so that all these signals are emitted by the corresponding transducer with the same total delay;

[0163] - Distortion: The filter and its characteristics (type, slope, etc.) are selected to limit the mechanical distortion rate of the transducer as much as possible and eliminate phase and time distortion;

[0164] - Bandwidth: There is no bandwidth limit;

[0165] - The distribution of the frequency bands assigned to each transducer depends on the trade-off between the array directivity, distortion, and the weight of the mobile device;

[0166] - Compression: Do not apply any dynamic range limit;

[0167] - Directivity: Perform on-axis and off-axis control of the directivity;

[0168] - Sampling: During digital processing, perform maximum oversampling; reverse the polarity of the transducer so that the impulse response is positive;

[0169] - Reference point: From the moment when the self-signal (from the front end of the speaker) is emitted, the phase and time curves are straight lines;

[0170] - The equalization condition is selected to linearize the frequency response amplitude curve as much as possible.

[0171] In addition, there are other feasible examples.

[0172] The present invention is not limited to the above and illustrated embodiments in any way, and those skilled in the art should understand how to deform it in any way according to their ideas.

Claims

1. A method for converting an audio signal of an electroacoustic transducer, characterized in that, According to the typical configuration scheme selected by the control module, the audio signal is corrected in a combined manner of multiple audio signal characteristics to provide specific characteristics to the audio signal, wherein the characteristics of the audio signal include: gain; phase; time; distortion; bandwidth; speaker bandwidth distribution; dynamic compression / expansion; directivity; sampling; absolute phase corresponding to the electrical polarity of the speaker group under impulse response; offset of the reference point when all frequencies are in phase, wherein the control module automatically adjusts the selection of the typical configuration scheme according to the information of the predetermined music type of the music track, and the selected typical configuration scheme includes the settings of the electroacoustic transducer for the characteristics of the audio signal.

2. The method according to claim 1, characterized in that, The audio signal is subjected to conversion through one or more steps, and the one or more steps include at least one correction operation of linearizing the audio signal to match the recorded data, and a correction operation of imparting the audio signal according to the selected typical configuration scheme.

3. The method according to claim 1 or 2, characterized in that, The conversion of the audio signal is implemented by a processor according to a digital method.

4. The method according to claim 1 or 2, characterized in that, The conversion of the audio signal is implemented by electrical and / or electronic components according to an analog method.

5. The method according to claim 1 or 2, characterized in that, The conversion of the audio signal is implemented according to one or more mechanical methods of converting using a tuning structure, an acoustic lens, and / or geometric characteristics of a device for converting the audio signal of the electroacoustic transducer.

6. The method according to claim 1 or 2, characterized in that, The control module is manually operated by the user.

7. The method according to claim 1 or 2, characterized in that, The control module is automatically adjusted according to the information for identifying the audio signal and the information for identifying the typical configuration scheme included in the remote service.

8. The method according to claim 1 or 2, characterized in that, The control module is automatically adjusted according to the user preferences identified by the device for converting the audio signal of the electroacoustic transducer.

9. A device for converting an audio signal of an electroacoustic transducer (3), characterized in that, The device is used to correct the audio signal in a combined manner of multiple audio signal characteristics according to the typical configuration scheme selected by the control module to endow the audio signal with specific characteristics, wherein the characteristics of the audio signal include: gain; phase; time; distortion; bandwidth; speaker bandwidth distribution; dynamic compression / expansion; directivity; sampling; absolute phase corresponding to the electrical polarity of the speaker group under impulse response; offset of the reference point when all frequencies are in phase, wherein the control module automatically adjusts the selection of the typical configuration scheme according to the information of the predetermined music type of the music track, and the selected typical configuration scheme includes the settings of the electroacoustic transducer for the characteristics of the audio signal.

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