Signal transmission system, signal conditioning module and signal conditioning method thereof

Abstract

Signal transmission systems, signal conditioning modules, and signal conditioning methods thereof are disclosed herein. In one embodiment, a signal transmission system includes a signal processing unit, an amplifier, and a transmitter. The signal processing unit can determine a delta limit based on a system parameter and a representative value of one or more samples of a conditioned signal, samples of a source signal, or a combination thereof. The signal processing unit can further determine a ratio of the delta limit to a difference between a sample of the source signal and a previous sample of the source signal, and apply a low pass filter to the source signal to obtain a filtered signal, where filter coefficients are generated based on the ratio. The amplifier can generate an amplified signal based on the filtered signal, and the transmitter can transmit an output signal based on the amplified signal.

Description

Technical Field

[0001] This disclosure is generally related to signal transmission systems. For example, several embodiments of the invention are generally related to signal conditioning modules for efficiently controlling current and power consumption in signal transmission systems with capacitive loads. Background Technology

[0002] Capacitive loads, such as piezoelectric loudspeakers, are widely used in various electronic devices for sound generation. These loads are characterized by their unique electrical properties; specifically, their impedance varies with frequency. As the frequency of the input signal increases, the impedance of a capacitive load typically decreases, which presents a potential challenge in driving these loads efficiently and safely. One such challenge is that signals at higher frequencies can cause current overload. Summary of the Invention

[0003] This disclosure relates to an adaptive current limiter for capacitive loads in signal transmission systems. For example, several embodiments of the invention relate to signal transmission systems with capacitive loads (e.g., piezoelectric loudspeakers) characterized by decreasing impedance with increasing frequency. This characteristic can pose a challenge to driving capacitive loads efficiently and safely, especially at higher frequencies where current overload can occur.

[0004] Therefore, to address these challenges, several embodiments of the present invention include a signal driver with signal conditioning modules configured to dynamically adjust the source signal to maintain the current in the signal transmission system within desired limits. This can be achieved by accurately estimating the current, efficiently determining the coefficients of a low-pass filter, adaptively applying the filter to the source signal, and gatekeeping individual signal samples. In some embodiments, the signal conditioning module includes a dynamically adjusted filter, a differential control block, and a limit / coefficient generator. The limit / coefficient generator can be configured to set the coefficients for the dynamically adjusted filter and the differential limits for the differential control block based at least in part on a number of system parameters (e.g., rated voltage, rated current, and rated power). Therefore, the present invention is expected to provide a balance between signal fidelity, current limiting, and computational efficiency when driving capacitive loads. Attached Figure Description

[0005] The various aspects of this disclosure are better understood with reference to the following accompanying drawings. The components in the drawings are not necessarily to scale. Rather, they are provided to clearly illustrate the principles of this disclosure. The drawings should not be construed as limiting this disclosure to the specific embodiments shown, but are provided for explanation and understanding.

[0006] Figure 1 Show the impedance or admittance characteristics of a piezoelectric loudspeaker at different frequencies;

[0007] Figure 2 This is a block diagram of a signal transmission system configured according to various embodiments of the present invention;

[0008] Figure 3 A block diagram of a digital signal processing unit configured according to various embodiments of the present invention;

[0009] Figure 4 A graph illustrating the differential limiting curves for different rated voltages and rated power according to various embodiments of the present invention;

[0010] Figure 5 To illustrate the frequency response curves of a first-order finite impulse response low-pass filter according to various embodiments of the present invention;

[0011] Figure 6 To illustrate the frequency response curves of a first-order infinite impulse response low-pass filter configured according to various embodiments of the present invention;

[0012] Figure 7 A flowchart illustrating a method for processing a signal transmitted via a capacitive load, according to various embodiments of the present invention;

[0013] Figure 8 A set of three vertically aligned graphs illustrating different states of signal processing for capacitive loads according to various embodiments of the present invention;

[0014] Figure 9 The following are shown: (i) a first spectrogram of an audio signal processed using a dynamically adjusted filter according to various embodiments of the present invention, and (ii) a second spectrogram of the same audio signal processed using a fixed bandwidth limiter.

[0015] Explanation of symbols in the attached drawings:

[0016] 102, 104. Impedance characteristic curves; 106. Admittance characteristic curves; 200. Signal transmission system; 210. Signal driver; 220. Digital signal processing unit; 230. Preprocessing module; 240. Signal conditioning module; 250. Amplifier; 260. Electromagnetic interference filter; 270. Transmitter; 275. Capacitive load; 295. Signal content; 332. Signal enhancement block; 334. Signal upsampling block; 336. Saturation control block; 342, Dynamic adjustment filter; 344, Differential control block; 346, Limit / coefficient generator; 348, System parameter block; 494, 596, 698, 891, 892, 893, 997, 999, Curves; 780, Method; 781, 782, 783, 784, 785, 786, 787, 788, 789, 790, 791, 792, Blocks; Cpz, Capacitor. Detailed Implementation

[0017] In the following description, specific details are set forth to provide a thorough understanding of the nature of the invention. However, those skilled in the art will recognize that the systems, apparatuses, and techniques described herein can be practiced without one or more of the specific details set forth herein or using other methods, components, materials, etc.

[0018] Throughout this specification, references to "example" or "embodiment" mean that a particular feature, structure, or characteristic described in connection with that example or embodiment is included in at least one example or embodiment of the invention. Therefore, the use of the phrases "by way of example," "as an example," or "embodiment" herein does not necessarily refer to the same example or embodiment, and is not necessarily limited to the specific example or embodiment discussed. Furthermore, the features, structures, or characteristics of the invention described herein can be combined in any suitable manner to provide other examples or embodiments of the invention.

[0019] A. Overview

[0020] Many signal transmission systems utilize capacitive loads, such as piezoelectric loudspeakers, which are characterized by decreasing impedance as frequency increases. For example, Figure 1Two impedance characteristic curves, 102 and 104, and one admittance characteristic curve, 106, are shown for a piezoelectric loudspeaker at different frequencies. Impedance characteristic curve 102 shows the impedance response on a logarithmic scale, where the impedance decreases with increasing frequency. Impedance characteristic curve 104 shows the impedance response on a semi-logarithmic scale, again demonstrating how the impedance decreases with increasing frequency. Admittance characteristic curve 106 shows the admittance response on a linear scale, illustrating how the admittance increases with increasing frequency. These impedance and admittance characteristics of capacitive loads pose a challenge to driving such loads efficiently and safely, especially at higher frequencies where current overload can occur.

[0021] One approach to this problem involves using a series resistor as a current limiter. However, this method introduces (a) unwanted heat dissipation and (b) constant signal attenuation at higher frequencies, resulting in reduced signal fidelity and undesirable power dissipation. Another approach involves using a fixed low-pass filter to reduce high-frequency content. While this avoids unwanted heat dissipation, it still introduces constant attenuation in the high-frequency region, degrading signal quality and making such methods unsuitable for high-fidelity reproduction. Another approach involves (i) estimating the current in the frequency domain, (ii) dynamically determining a filter with the desired cutoff frequency and roll-off slope, (iii) applying the filter to the signal in the frequency domain, and (iv) reconstructing the signal back to the time domain. Such methods typically require buffering samples and performing complex mathematical operations (e.g., Fast Fourier Transform (FFT) and iterative algorithms for polynomial root finding). These methods therefore introduce considerable processing latency and require complex hardware implementations, making them less suitable for applications where low latency and reduced computational complexity are critical.

[0022] In contrast, several embodiments of the present invention relate to signal transmission systems with capacitive loads (e.g., piezoelectric loudspeakers) driven by signal drivers that implement signal conditioning modules configured to dynamically adjust the source signal. More specifically, the signal conditioning module configured according to various embodiments of the present invention may include a dynamically adjusted filter, a differential control block, and a limit / coefficient generator. The limit / coefficient generator may set (i) coefficients for the dynamically adjusted filter and (ii) differential limits for the differential control block based on multiple system parameters (e.g., rated voltage, rated current, and rated power). In operation, the signal conditioning module can maintain the current consumption in the signal transmission system within desired limits by accurately estimating the current, efficiently determining the coefficients of the low-pass filter, adaptively applying the low-pass filter to the source signal, and gate each signal sample.

[0023] Therefore, the present invention is expected to offer several advantages over various other methods described above for controlling current consumption when driving capacitive loads. For example, unlike series resistors that introduce constant attenuation and heat dissipation, or fixed low-pass filters that compromise signal quality, the present invention dynamically adapts to signal characteristics to control current consumption. This adaptive approach of the present invention allows signal fidelity to be maintained where possible, while still preventing current overload and unwanted heat dissipation. Furthermore, compared to methods requiring complex frequency domain operations, the present invention operates in the time domain. Therefore, the present invention does not require signal buffering or the use of complex mathematical operations. In fact, the present invention utilizes efficient, low-complexity algorithms. Thus, the present invention is expected to achieve low latency and reduced computational complexity, and further is expected to enable implementation using relatively simple hardware. In other words, the present invention is expected to provide a balance between signal fidelity, current limitation, and computational efficiency when driving capacitive loads.

[0024] B. Selected embodiments of adaptive current limiters and related systems, devices, and methods for capacitive loads.

[0025] Figure 2 This is a block diagram of a signal transmission system 200 configured according to various embodiments of the present invention. As shown, the signal transmission system 200 includes a signal driver 210 and a transmitter 270. The signal driver 210 includes a digital signal processing unit 220 and an amplifier 250. The digital signal processing unit 220 includes a preprocessing module 230 and a signal conditioning module 240. In some embodiments, the signal transmission system 200 may further include an electromagnetic interference (EMI) filter 260. In other embodiments, the EMI filter 260 may be omitted.

[0026] The signal transmitting system 200 is configured to receive signal content 295 as input. In some embodiments, the signal content 295 may include an audio signal. As a specific example, the signal content 295 may include an audio signal having an audio bandwidth of about 60 Hz to 20 kHz, a sampling rate of 44.1 kHz or 48 kHz, and a bit depth of 16, 24, or 32 bits. The signal content 295 may originate from various sources, such as storage devices or streaming devices.

[0027] The digital signal processing unit 220 of the signal driver 210 processes the received signal content 295. In some embodiments, the preprocessing module 230 of the digital signal processing unit 220 performs initial signal processing operations. Such operations may include audio enhancement techniques, bass boost, equalization (EQ), upsampling, and / or saturation control. After the preprocessing module 230, the signal conditioning module 240 of the digital signal processing unit 220 dynamically modifies / adjusts the signal content to produce a conditioned signal. See below. Figure 3In more detail, the signal conditioning module 240 is configured to manipulate the signal content in a manner that reduces or controls current consumption in the signal transmission system 200.

[0028] Following digital signal processing, the amplifier 250 of the signal driver 210 amplifies the regulated signal output from the signal conditioning module 240. In some embodiments, the amplifier 250 is a Class D or Class AB amplifier with specified rated voltage, current, and power (VIP). The amplified signal output from the amplifier 250 is then provided to an EMI filter 260 (if present) or to a transmitter 270.

[0029] When included, the EMI filter 260 may be designed to reduce, minimize, or eliminate electromagnetic interference (e.g., noise). In some embodiments, the EMI filter 260 includes an LC filter comprising one or more inductors and one or more capacitors configured in a specific configuration to filter out unwanted frequencies / noise. In the illustrated embodiment, each inductor of the EMI filter 260 has an inductance L, and each capacitor of the EMI filter 260 includes a capacitance C. The inductors in the EMI filter 260 may have a specified rated current. Alternatively or additionally, the EMI filter 260 may include a ferrite bead filter.

[0030] The output of EMI filter 260 (if present) or an amplified signal directly from amplifier 250 (when EMI filter 260 is omitted) may be provided to transmitter 270. Transmitter 270 is then configured to transmit an output signal. As shown, transmitter 270 includes a capacitive load 275. In some embodiments, capacitive load 275 includes a piezoelectric speaker, which may have a specified rated voltage and rated power. The capacitive load 275 in... Figure 2 The capacitor Cpz is represented in this context. In some embodiments, the transmitter 270 may exhibit an increasing admittance (and / or a decreasing impedance) as the frequency of the amplified signal increases.

[0031] As discussed in more detail below, the signal transmitting system 200 is designed to process and transmit signals while managing electromagnetic interference and / or adapting to the impedance / admittance characteristics of the capacitive load 275. More specifically, the digital signal processing unit 220 of the signal driver 210—particularly the signal conditioning module 240—enables the signal transmitting system 200 to efficiently process signal content across a range of frequencies and amplitudes, while addressing the challenges associated with driving capacitive loads, especially at higher frequencies where current overloads can occur.

[0032] Figure 3This is a block diagram of a digital signal processing unit 320 configured according to various embodiments of the present invention. For example, the digital signal processing unit 320 may be a digital signal processor for regulating signals of a capacitive load. The digital signal processing unit 320 may be... Figure 2 Examples of digital signal processing units 220, or other digital signal processing units configured according to various embodiments of the present invention.

[0033] As shown, the digital signal processing unit 320 includes a preprocessing module 330 and a signal conditioning module 340. Figure 3 The components of the preprocessing module 330 are shown using dashed / dashed boxes and corresponding arrows, while the components of the signal conditioning module 340 are shown using solid boxes and corresponding arrows. In the illustrated embodiment, the preprocessing module 330 includes a signal enhancement block 332, a signal upsampling block 334, and a saturation control block 336. The signal conditioning module 340 includes a dynamic adjustment filter 342, a differential control block 344, a limit / coefficient generator 346, and a system parameter block 348. As discussed in more detail below, the components of the signal conditioning module 340 are configured to manipulate the source signal x to reduce the corresponding signal transmission system (e.g., Figure 2 The current consumption of the signal transmission system 200.

[0034] First, referring to the preprocessing module 330, the signal enhancement block 332 receives the signal content x0 and outputs the enhanced signal content xe. In some embodiments, the signal enhancement block 332 performs audio enhancement techniques on the input signal content x0. These techniques may include bass boost, equalization (EQ), or other audio processing operations to improve the quality or characteristics of the signal content x0.

[0035] Furthermore, the signal upsampling block 334 receives the enhanced signal content xe from the signal enhancement block 332 and increases the sampling rate to generate the upsampled signal content xu. In some embodiments, the signal upsampling block 334 upsamples the signal to at least 128 kHz. This higher sampling rate is expected to enable more accurate current estimation in subsequent processing stages. As a specific example, the signal upsampling block 334 may quadruple the sampling rate of the enhanced signal content xe, such as from 48 kHz to 192 kHz. The upsampling performed by the signal upsampling block 334 ensures that the maximum gradient of the upsampled signal content xu is greater than or equal to the maximum gradient of the enhanced signal content xe.

[0036] Saturation control block 336 may receive an upsampled signal xu from signal upsampling block 334 and use it to generate a source signal x. In some embodiments, saturation control block 336 limits (e.g., compresses) the upsampled content xu to an effective dynamic range (e.g., -3 dB) based on system parameters received from system parameter block 348. For example, an overvoltage observed in the upsampled signal xu at saturation control block 336 may be attributed to overfeeding, enhancement performed by signal enhancement block 332, and / or upsampling performed by signal upsampling block 334. Continuing this example, saturation control block 336 may limit the output voltage of the source signal x to less than or equal to (amplifier, such as...) Figure 2 The amplifier 250) supplies power to the source signal x while maintaining the fidelity of the lower amplitude signal content. As another example, the saturation control block 336 can limit the output voltage of the source signal x to be less than or equal to a voltage limit Vx (e.g., the amplifier 250). Figure 2 The power supply voltage of the amplifier 250) and the capacitive load (e.g., Figure 2 The rated voltage of the capacitive load (275) and the LC filter (e.g., Figure 2 The EMI filter 260 uses capacitors with rated voltages at the minimum rated value to maintain fidelity of the low-amplitude signal content. As a specific example, saturation control block 336 can limit the output voltage of the source signal x to less than or equal to 12 V. In some embodiments, the effective dynamic range can be the ratio of the voltage limit Vx to the full voltage Vf. As a specific example, assuming the voltage limit Vx is 12 V and the full voltage Vf is 15.4 V, the effective dynamic range based on these system parameters could be ±12 V / 15.4 V or ±0.779 (e.g., -2.16 dB). Figure 3 As shown, the saturation control block 336 outputs the source signal x to the dynamic adjustment filter 342 and the limit / coefficient generator 346 of the signal conditioning module 340.

[0037] Referring now to signal conditioning module 340, system parameter block 348 contains a signal transmission system (e.g., corresponding to digital signal processing unit 320) Figure 2The ratings and specifications of the signal transmitting system 200 are as follows. These ratings and specifications may depend on the components of the signal transmitting system. For example, the ratings and specifications may depend on the ratings and specifications of components placed in series with the transmitter (e.g., transmitter 270) of the signal transmitting system. The ratings and specifications contained in the system parameter block 348 may include rated voltage, current, power (VIP), saturation current, sampling rate, load capacitance, etc. For example, system parameters may include full voltage Vf (e.g., 15.4 V), voltage limit Vx, current limit Ix, power limit Px, combined (or load) capacitance C, saturation current, sampling rate Fs (e.g., 192 kHz), and other parameters.

[0038] As discussed above, the voltage limit Vx can be equivalent to the supply voltage (e.g., 12 V) or equivalent to (i) the amplifier (e.g., Figure 2 (ii) the supply voltage of the amplifier 250), and the capacitive load (e.g., Figure 2 (iii) the rated voltage of the capacitive load 275), and the capacitors of the LC filter (e.g., Figure 2 The minimum rated voltage of the EMI filter 260 (LC filter). As a specific example, the voltage limit Vx can be 12 V.

[0039] The current limit Ix can be less than or equal to the transmitter of the corresponding signal transmitting system (e.g., Figure 2 The minimum rated current and saturation current of any component placed in series with the transmitter 270. For example, the current limit Ix may be equivalent to (a) an amplifier (e.g., Figure 2 (a) amplifier 250), (b) (e.g., Figure 2 The EMI filter 260 has an inductor, and (c) a capacitive load (e.g., Figure 2 The minimum rated current and saturation current of the capacitive load (275). As a specific example, the current limit Ix can be 3 A.

[0040] Similarly, the power limit Px can be equivalent to an amplifier (e.g., Figure 2 The rated power of the amplifier 250 and the minimum rated power of the corresponding signal transmitter (e.g., transmitter 270) in the signal transmission system. The load capacitance C may be equivalent to a capacitive load (e.g., Figure 2 The capacitor (275) of the capacitive load, such as the EMI filter omitted in the corresponding signal transmission system (e.g., Figure 2 When using an EMI filter 260, the load capacitance C can be equivalent to a combined capacitor. For example, the load capacitance C can be equivalent to (i) the capacitance of a capacitive load (e.g., Figure 2 (ii) capacitive load 275) and (e.g., Figure 2 The EMI filter 260 is a combination of capacitors of an LC filter, such as in embodiments that include the use of an LC filter as an EMI filter.

[0041] like Figure 3 As shown, system parameter block 348 is configured to provide system parameters to saturation control block 336 of preprocessing module 330 (e.g., to enable saturation control block 336 to limit the source signal x within its effective dynamic range). Additionally, system parameter block 348 is configured to provide system parameters to limit / coefficient generator 346.

[0042] The limit / coefficient generator 346 determines the appropriate filter coefficients for the dynamic adjustment filter 342 and the appropriate differential limits for the differential control block 344. The filter coefficients and differential limits generated by the limit / coefficient generator 346 can be based on the system parameters received from the system parameter block 348, the characteristics of the source signal x, the characteristics of the filtered signal yf output from the dynamic adjustment filter 342, and / or the characteristics of the adjusted signal y output from the differential control block 344.

[0043] 1. Differential Limitation

[0044] The limit / coefficient generator 346 can determine the differential limit dx_lmt based on both current and power constraints, thereby selecting more restrictive options for both. For example, the differential limit generated by the limit / coefficient generator 346 can be based on system parameters received from the system parameter block 348, the characteristics of the source signal x output from the self-saturation control block 336, and / or the characteristics of the regulated signal y output from the differential control block 344. Specifically, the current of a capacitive load is proportional to the first derivative of its voltage, as shown in Equation 1 below, where C, Fs, and Vf represent the load capacitance, sampling rate, and full voltage provided by the system parameter block 348, respectively:

[0045] Equation 1:

[0046]

[0047] In Equation 1 above, dx represents the difference between a sample of the source signal x and a previous sample of the source signal x, as shown in Equation 2 below:

[0048] Equation 2:

[0049]

[0050] In some embodiments, the difference dx can be the peak-to-peak difference of the signal content at the Nyquist frequency.

[0051] Given a current limit Ix (e.g., received from system parameter block 348) and a differential limit dxi This can therefore be provided by the following equation 3:

[0052] Equation 3:

[0053]

[0054] The power P is provided by the following equation 4, where xr represents a representative value (0~1) of a sample of the source signal x output by the saturation control block 336, a sample of the regulated signal y output by the differential control block 344, or a combination thereof:

[0055] Equation 4:

[0056]

[0057] (For example, the differential limit dx of the given power limit Px received from system parameter block 348) p Therefore, it can be provided by the following equation 5, where xr is a representative value of the worst signal between (i) the source signal x output by the self-saturation control block 336 and (ii) the adjusted signal y output by the differential control block 344:

[0058] Equation 5:

[0059]

[0060] The representative value xr can be provided by the following equation 6, where Let be the instantaneous value of the source signal x, and let yr be the representative value of the regulated signal y output by the differential control block 344:

[0061] Equation 6:

[0062]

[0063] According to Equation 6 above, when the instantaneous value When the representative value xr of the adjusted signal y is greater than (or equal to) the representative value yr, the representative value xr is the instantaneous value of the source signal x. Otherwise, the representative value yr of the adjusted signal y is used as the representative value xr in Equation 5 above to determine the differential limit dx of the given power limit Px. p .

[0064] In some embodiments, the representative value yr of the regulated signal y output by the differential control block 344 can be initially set to zero and then updated to be equal to the instantaneous value of the first sample of the regulated signal y. Thereafter, the representative value yr of the regulated signal y can be adjusted over time. For example, as the regulated signal y output by the differential control block 344 changes, when the instantaneous value of the regulated signal y... When the value of the representative value yr of the adjusted signal y is greater than the current value of the representative value yr, the representative value yr can be updated to be equal to the instantaneous value of the adjusted signal y. (For example, if yr < Then update yr so that yr = On the other hand, it is assumed that the current value of the representative quantity yr remains greater than (or equal to) the instantaneous value of the adjusted signal y. The representative value yr of the adjusted signal y can (a) remain for a preset time period (or the existing number of samples), and then (b) be gradually adjusted / updated. As a specific example, assume that the current value of the representative value yr of the adjusted signal y remains greater than (or equal to) the instantaneous value of the adjusted signal y over a longer period of time. The representative value yr can be maintained for a preset time period tHold (e.g., 1 ms), which corresponds to a preset number of samples nHold for the adjusted signal y at a given sampling rate (e.g., 48 samples at 48 kHz). Then, the representative value yr can be updated according to the following equations 7-10, such that (a) the representative value yr of the adjusted signal y is adapted to the maximum instantaneous value yr1 of the adjusted signal y within the preset number of samples nHold, and (b) the representative value yr is adjusted according to a preset time constant Tc (e.g., 1 ms), which corresponds to a preset number of samples nTc at a given sampling rate (e.g., 48 samples at 48 kHz).

[0065] Equation 7:

[0066]

[0067] Equation 8:

[0068]

[0069] Equation 9:

[0070]

[0071] Equation 10:

[0072]

[0073] As shown by the following Equation 11, the limit / coefficient generator 346 can be configured to set the differential limit dx_lmt to a differential limit dx equivalent to (i) the given current limit Ix. i and (ii) the differential limit dx given the power limit Px p Minimum value:

[0074] Equation 11:

[0075] dx_lmt=min(dx i ,dx p )

[0076] In other embodiments, the limit / coefficient generator 346 may set the differential limit dx_lmt to a differential limit dx equivalent to the per-current limit. i Or the differential limit dx per power limit p (For example, it may not be the minimum of the two).

[0077] In some embodiments, the limit / coefficient generator 346 may use a lookup table (LUT) to determine the differential limit dx_lmt. For example, the limit / coefficient generator 346 may initialize the differential limit value dx based on the source signal x or based on a representative value xr. p Then, during sample processing, a LUT is used to determine the appropriate difference limit dx_lmt.

[0078] Figure 4 Curve 494 is used to illustrate the differential limiting curves for different rated voltage, current, and power (VIP) at a series of representative values ​​xr according to various embodiments of the present invention. As shown, the x-axis of curve 494 represents the full representative value xr from 0 to 1 (which may be the instantaneous value of the source signal x or the representative value yr of the regulated signal y output by the differential control block 344), while the y-axis represents the full differential limiting dx. i dx p value.

[0079] Curve 494 graphically represents multiple curves corresponding to different rated voltage, current, and power (VIP). For example, the vertical dashed line is positioned at approximately 0.779 along the x-axis of curve 494. This vertical dashed line corresponds to a voltage limit Vx of 12 V. More specifically, assuming the voltage limit Vx is 12 V and the full voltage Vf is 15.4 V, the vertical dashed line on the x-axis of curve 494 is positioned at the quotient of the voltage limit Vx divided by the full voltage Vf, which corresponds to the voltage limit of Vx divided by the full voltage Vf. Figure 3 The saturation control block 336 of the preprocessing module 330 imposes a voltage limit on the sampled content xu to ensure that the source signal x output by the self-saturation control block 336 is within the effective dynamic range of the signal conditioning module 340.

[0080] Curve 494 further includes two horizontal lines positioned at approximately 0.211 and 0.070 on the y-axis. The horizontal line positioned at approximately 0.211 on the y-axis corresponds to a current limit of 3 A Ix, and the horizontal line positioned at approximately 0.070 on the y-axis corresponds to a current limit of 1 A. More specifically, assuming a current limit Ix of 3 A, a load capacitance C of 4.8 µF, a sampling rate Fs of 192 kHz, and a full voltage Vf of 15.4 V, Equation 3 above can be used to determine... Figure 4 The position of the top horizontal line on the y-axis at approximately 0.211 in curve 494. Similarly, assuming a current limit Ix of 1 A, a load capacitance C of 4.8 µF, a sampling rate Fs of 192 kHz, and a full voltage Vf of 15.4 V, Equation 3 above can be used to determine... Figure 4 The position of the top horizontal line on the y-axis at approximately 0.070 in curve 494. These horizontal lines are referred to in this paper as the current limiting curve.

[0081] Curve 494 further includes power limit curves corresponding to power limits Px of 30 W, 20 W, 10 W, and 5 W. Each of these curves shows the differential limit dx for the respective rated power. p How does the value change with the representative value xr? For example, the power limitation curves in these figures show that as the representative value xr increases, the corresponding differential limitation dx... p Decrease. This relationship reflects the inverse correlation between the representative value xr and the allowable amplitude variation (difference) between consecutive samples of the regulated signal y output by the differential control block 344, in order to maintain a given power limit. Equation 5 above can be used to determine... Figure 4 The values ​​of each in the power limitation curve shown in the figure.

[0082] As discussed above, the limit / coefficient generator 346 can set the differential limit dx_lmt at any given point in curve 494 corresponding to the minimum of the following: (a) the differential limit dx corresponding to the applicable current limit curve. i The value, and (b) the differential limit dx corresponding to the applicable power limit curve. p For example, at a lower representative value xr, the applicable current limiting curve can be a decision factor. Therefore, at a lower representative value xr, the limit / coefficient generator 346 can set the differential limit dx_lmt to be equivalent to the corresponding dx. iThe power limiting curve may intersect with the current limiting curve at a higher representative value xr, thus potentially becoming a determining factor for the differential limiting dx_lmt at the higher representative value xr. In other words, the limiting / coefficient generator 346 can set the differential limiting dx_lmt to be equivalent to (a) the corresponding differential limiting dx at a lower representative value xr. i The value, and (b) the corresponding difference constraint dx for a higher representative value xr. p value.

[0083] In some embodiments, the limit / coefficient generator 346 may use curve 494 (or the relationship shown in curve 494) to dynamically adjust the differential limit dx_lmt based on (i) a representative value xr and (ii) system parameters received from the system parameter block 348. For example, the limit / coefficient generator 346 may calculate the differential limit dx per current limit based on Equation 3. i Furthermore, the differential limit dx per power limit is calculated based on Equation 5. p Then, the minimum of these two values ​​is selected as the final difference limit dx_lmt, as shown in Equation 11. In these and other embodiments, the limit / coefficient generator 346 can use the relationship shown in curve 494 to initialize the difference limit dx based on the representative value xr. p The value is then used, and a lookup table (LUT) is used to determine the appropriate differential limit dx_lmt during sample processing. This method allows for efficient and real-time adjustment of the differential limit based on (i) changes in characteristics of the source signal x, (ii) changes in characteristics of the regulated signal y, and / or (iii) system parameters received from system parameter block 348. As discussed in more detail below, after the limit / coefficient generator 346 determines the appropriate differential limit dx_lmt, the limit / coefficient generator 346 can provide the differential limit dx_lmt to the differential control block 344, which can use its limitation relative to the previous samples of the regulated signal y to the filtered signal yf output from the dynamic adjustment filter 342, as discussed in more detail below.

[0084] Figure 4 Curve 494 also shows how different system parameters affect the differential limit dx. i dx p And dx_lmt. For example, curve 494 shows a higher current limit Ix corresponding to a higher level current limit curve, thus potentially allowing a larger difference value dx_lmt under multiple (e.g., lower) representative values ​​xr. Similarly, a higher power level can shift the corresponding power limit curve upward, thus potentially allowing a larger difference value dx_lmt under multiple (e.g., higher) representative values ​​xr.

[0085] 2. Filter coefficients

[0086] like Figure 3 As shown, the dynamic adjustment filter 342 receives the source signal x output by the saturation control block 336 and filters it to generate a filtered signal yf based on the filter coefficients determined by the limit / coefficient generator 346 of the signal conditioning module 340. In other words, the dynamic adjustment filter 342 converts the source signal x into the filtered signal yf based on the filter coefficients provided by the limit / coefficient generator 346.

[0087] In some embodiments, the dynamically adjusted filter 342 may be a first-order finite impulse response (FIR) filter. In these embodiments, the filtered signal yf can be modeled using the following Equation 12:

[0088] Equation 12:

[0089]

[0090] In Equation 12 above, B0 and B1 are coefficients that can be determined and provided by the limit / coefficient generator 346 of the signal conditioning module 340.

[0091] The following equation 13 models the filtered signal yf provided by the FIR filter in the frequency domain:

[0092] Equation 13:

[0093]

[0094] Therefore, the transformation function of the FIR filter is shown by the following equation 14:

[0095] Equation 14:

[0096]

[0097] When the difference dx between a sample of the source signal x and a previous sample of the source signal x exceeds the difference limit dx_lmt determined by the limit / coefficient generator 346 (as discussed above), it can be loosely assumed that the desired gain g at the Nyquist frequency (e.g., half the sampling rate Fs) is given by the ratio dx_lmt / dx. Therefore, given DC (z -1 The desired unit gain and Nyquist frequency (z) under condition = 1) -1 The desired gain g at DC = -1) can be obtained by establishing the following equations 15 (representing the unit gain at DC) and 16 (representing the desired gain g at the Nyquist frequency) to solve for the filter coefficients B0 and B1:

[0098] Equation 15:

[0099]

[0100] Equation 16:

[0101] g

[0102] Solving equations 15 and 16 simultaneously yields equations 17 and 18. Equations 17 and 18 can be used by the constraint / coefficient generator 346 to determine coefficients c (i.e., B0 and B1) and provide these coefficients to the dynamic adjustment filter 342 to convert the source signal x into the filtered signal yf.

[0103] Equation 17:

[0104] B0 = (1 + g) / 2

[0105] Equation 18:

[0106] B1=(1-g) / 2

[0107] In some embodiments, the limit / coefficient generator 346 may use a lookup table (LUT) to determine coefficients B0 and / or B1.

[0108] Coefficients B0 and B1 ensure that the first-order FIR filter maintains the unit gain under DC conditions while achieving the desired gain g at the Nyquist frequency. Furthermore, by adjusting the gain g based on the ratio of the differential limit dx_lmt to the observed difference dx, the filter can dynamically adapt its frequency response to the characteristics of the source signal x, thereby effectively limiting rapid changes that could lead to excessive current in capacitive loads. Moreover, since the differential limit dx_lmt is based on (i) the system parameters received from system parameter block 348, (ii) the characteristics of the source signal x output from self-saturation control block 336, and (iii) the characteristics of the regulated signal y output from differential control block 344 (as discussed in more detail above and below), the filter coefficients of the first-order FIR filter of the dynamically adjusted filter 342 can similarly be based on the system parameters received from system parameter block 348, the characteristics of the source signal x output from self-saturation control block 336, and the characteristics of the regulated signal y output from differential control block 344.

[0109] Figure 5 Curves 596 illustrate the frequency response of first-order finite impulse response (FIR) low-pass filters with different coefficient sets according to various embodiments of the present invention. As discussed above, in some embodiments, Figure 3 The dynamically adjustable filter 342 can be implemented as a first-order FIR filter. Figure 5Curve 596 illustrates how the frequency response of a first-order FIR filter changes as the coefficients are adjusted. The x-axis of curve 596 represents frequency in Hz, while the y-axis represents gain on a linear scale from 0 to 1. The curves in curve 596 correspond to different sets of filter coefficients, thus demonstrating the flexibility of a first-order FIR filter in shaping its frequency response based on selected coefficient values.

[0110] In some embodiments, the curve in curve 596 can span a range of gain responses from almost smooth to steep roll-off at higher frequencies. For example, one curve may represent a filter configuration with coefficients [B0, B1] = [0.95, 0.05], which can produce a relatively smooth frequency response. Another curve may represent a filter configuration with coefficients [B0, B1] = [0.50, 0.50], which can produce a stronger low-pass filtering effect with a steeper roll-off at higher frequencies.

[0111] As discussed above, Figure 3 The limiting / coefficient generator 346 can dynamically adjust these coefficients based on the characteristics of the source signal x, the characteristics of the regulated signal y, and the system parameters received from the system parameter block 348. This dynamic adjustment allows the dynamically adjusted filter 342 to adapt its frequency response in real time, thereby effectively controlling the signal content at different frequencies to manage current consumption in capacitive loads, while maintaining signal fidelity where possible.

[0112] In some embodiments, the ability to adjust the filter coefficients can provide a balance between preserving the signal content and limiting rapid changes in current that could lead to excessive current in capacitive loads. For example, when signal characteristics indicate a low risk of current overload, the coefficients can be set to produce a smoother frequency response, thus preserving more of the original signal content. Conversely, when signal characteristics indicate a high risk of current overload, the coefficients can be adjusted to produce a stronger low-pass filtering effect, thereby attenuating high-frequency components that could cause rapid current changes.

[0113] See again Figure 3 The dynamically adjusted filter 342 of the signal conditioning module 340 can alternatively be implemented as a first-order infinitesimal impulse response (IIR) filter. In these embodiments, the filtered signal yf can be modeled using the following equation 19:

[0114] Equation 19:

[0115]

[0116] In Equation 19 above, B0 and A1 are coefficients that can be determined and provided by the limit / coefficient generator 346 of the signal conditioning module 340.

[0117] The following equation 20 models the filtered signal yf provided by the IIR filter in the frequency domain:

[0118] Equation 20:

[0119]

[0120] Therefore, the transformation function of the IIR filter is shown by the following equation 21:

[0121] Equation 21:

[0122]

[0123] Similarly, when the difference dx between a sample of the source signal x and a previous sample of the source signal x exceeds the difference limit dx_lmt determined by the limit / coefficient generator 346 (as discussed above), it can be loosely assumed that the desired gain g at the Nyquist frequency (e.g., half the sampling rate Fs) is given by the ratio dx_lmt / dx. Therefore, given DC (z -1 The desired unit gain and Nyquist frequency (z) at =1) -1 The desired gain g at DC = -1) can be obtained by establishing the following equations 22 (representing the unit gain at DC) and 23 (representing the desired gain g at the Nyquist frequency) to solve for the filter coefficients B0 and A1:

[0124] Equation 22:

[0125] B0+A1=1

[0126] Equation 23:

[0127] B0=(1+A1)*g

[0128] Solving equations 22 and 23 simultaneously yields equations 24 and 25. Equations 24 and 25 can be used by the constraint / coefficient generator 346 to determine coefficients c (i.e., B0 and A1) and provide these coefficients to the dynamic adjustment filter 342 to convert the source signal x into the filtered signal yf.

[0129] Equation 24:

[0130] B0 = 2g / (1+g)

[0131] Equation 25:

[0132] A1 = (1-g) / (1+g)

[0133] In some embodiments, the limit / coefficient generator 346 may use a lookup table (LUT) to determine coefficients B0 and / or A1.

[0134] Coefficients B0 and A1 ensure that the first-order IIR filter maintains unit gain at DC frequency while achieving the desired gain g at the Nyquist frequency. Furthermore, by adjusting the gain g based on the ratio of the differential limit dx_lmt to the observed difference dx, the filter can dynamically adapt its frequency response to the characteristics of the source signal x, effectively limiting rapid changes that could lead to excessive current in capacitive loads. Moreover, since the differential limit dx_lmt is based on the system parameters received from system parameter block 348, the characteristics of the source signal x output from self-saturation control block 336, and the characteristics of the regulated signal y output from differential control block 344 (as discussed in more detail above and below), the filter coefficients of the first-order IIR filter of the dynamically adjusted filter 342 can similarly be based on the system parameters received from system parameter block 348, the characteristics of the source signal x output from self-saturation control block 336, and the characteristics of the regulated signal y output from differential control block 344.

[0135] Figure 6 Curve 698 illustrates the frequency response of a first-order infinite impulse response (IIR) low-pass filter with different coefficient sets according to various embodiments of the present invention. As discussed above, in some embodiments, Figure 3 The dynamically adjustable filter 342 can be implemented as a first-order IIR filter. Figure 6 Curve 698 illustrates how the frequency response of a first-order IIR filter changes as the coefficients are adjusted. The x-axis of curve 698 represents the frequency in Hz, while the y-axis represents the gain on a linear scale from 0 to 1. The curves in curve 698 correspond to different sets of filter coefficients, thus demonstrating the flexibility of adjusting the frequency response of a first-order IIR filter based on the selected coefficient values.

[0136] In some embodiments, the curves in curve 698 may span a range of gain responses from nearly smooth to steeply roll-off. For example, one curve may represent a filter configuration with coefficients [B0, A1] = [0.95, 0.05], which can produce a relatively smooth frequency response. Another curve may represent a filter configuration with coefficients [B0, A1] = [0.10, 0.90], which can produce a stronger low-pass filtering effect with a steeper roll-off.

[0137] As discussed above, Figure 3 The limiting / coefficient generator 346 can dynamically adjust these coefficients based on the characteristics of the source signal x, the characteristics of the regulated signal y, and the system parameters received from the system parameter block 348. This dynamic adjustment allows the dynamically adjusted filter 342 to adapt its frequency response in real time, thereby effectively controlling the signal content at different frequencies to manage current consumption in capacitive loads, while maintaining signal fidelity where possible.

[0138] In some embodiments, the ability to adjust the filter coefficients can provide a balance between preserving the signal content and limiting rapid changes in current that could lead to excessive current in capacitive loads. For example, when signal characteristics indicate a low risk of current overload, the coefficients can be set to produce a smoother frequency response, thus preserving more of the original signal content. Conversely, when signal characteristics indicate a high risk of current overload, the coefficients can be adjusted to produce a stronger low-pass filtering effect, thereby attenuating high-frequency components that could cause rapid current changes.

[0139] 3. Step / Differential Limiter

[0140] See again Figure 3The filtered signal yf output from the self-dynamically adjusted filter 342 is fed to the differential control block 344. The differential control block 344 may be a step limiter configured to limit inter-sample variations in the regulated signal y based on the differential limit dx_lmt received by the self-limiting / coefficient generator 346 (e.g., to help prevent rapid changes in the regulated signal y that could lead to excessive current or voltage in a coupled capacitive load). More specifically, the differential control block 344 is configured to limit the current sample of the filtered signal yf output from the self-dynamically adjusted filter 342 using the differential limit dx_lmt relative to the previous (e.g., immediately preceding) sample of the regulated signal y output from the self-differential control block 344. Specifically, in some embodiments, the differential control block 344 is configured to limit the difference df (e.g., df = yf[i] - y[i-1]) between (a) the current sample of the filtered signal yf and (b) the last sample of the adjusted signal y, to ensure that the difference remains less than or equal to the differential limit dx_lmt. For example, when the magnitude of the difference df between the current sample yf[i] of the filtered signal and the previous sample y[i-1] of the adjusted signal is less than or equal to the differential limit dx_lmt received from the limit / coefficient generator 346, the differential control block 344 may be configured to allow a sample of the filtered signal yf to pass through such that, for example, the current sample yf[i] of the adjusted signal output from the differential control block 344 is equivalent to the previous sample y[i-1] of the adjusted signal plus the difference df. In other embodiments, when the magnitude of the difference df between the current sample yf[i] of the filtered signal and the previous sample y[i-1] of the regulated signal is less than or equal to the difference limit dx_lmt received by the self-limiting / coefficient generator 346, the differential control block 344 may be configured to allow samples of the filtered signal yf with a unit gain g (e.g., g=1) to pass through. On the other hand, when the magnitude of the difference df between the current sample of the filtered signal yf and the previous sample of the regulated signal y is greater than the difference limit dx_lmt received by the self-limiting / coefficient generator 346, the difference control block 344 can be configured to limit the magnitude of the difference df to the difference limit dx_lmt while maintaining the sign s of the difference df (e.g., positive one (1) or negative one (-1)), such that, for example, the current sample yf[i] of the regulated signal output by the difference control block 344 is equivalent to the previous sample y[i-1] of the regulated signal plus the product of the sign s of the difference df and the difference limit dx_lmt (e.g., y[i] = y[i-1] + s * dx_lmt). In other embodiments, when the magnitude of the difference df between a sample of the filtered signal yf and a previous sample of the regulated signal yf is greater than the difference limit dx_lmt, the differential control block 344 may be configured to allow samples of the filtered signal yf with a gain g less than one (e.g., g = dx_lmt / df) to pass through.The gain g can be the desired gain at the Nyquist frequency.

[0141] In some embodiments, the output of the differential control block 344 is an adjusted signal y, which can be output from the signal conditioning module 340 of the digital signal processing unit 320 to the corresponding signal driver (e.g., ...). Figure 2 The amplifier of the signal driver 210 (e.g., Figure 2 (Amplifier 250). Furthermore, as Figure 2 As shown, the amplifier's output can be supplied to an EMI filter (e.g., Figure 2 EMI filter 260) and / or transmitter (e.g., Figure 2 The transmitter 270), to transmit via a capacitive load (e.g., Figure 2 The capacitive load 275) emits a related signal. Therefore, by Figure 3 The operation performed by the digital signal processing unit 320—particularly the signal conditioning module 340—is particularly beneficial when driving capacitive loads such as piezoelectric loudspeakers, which can exhibit challenging impedance characteristics at high frequencies.

[0142] 4. Related Methods

[0143] Figure 7 A flowchart illustrating a method 780 for processing a signal emitted via a capacitive load according to various embodiments of the present invention is provided. For example, method 780 can be used to regulate a signal emitted by a piezoelectric loudspeaker while (a) limiting excessive current and voltage at higher frequencies and (b) maintaining signal fidelity where possible. Method 780 is shown as a series of blocks 781-792. All or a subset of one or more of blocks 781-792 of method 780 can be generated by a signal transmitting system (e.g., Figure 2 The signal transmission system 200) is executed by various components or devices, such as a digital signal processing unit (e.g., Figure 2 Digital signal processing unit 220 Figure 3 Digital signal processing unit 320 Figure 3 The preprocessing module 330 and / or Figure 4 Signal conditioning module 340), amplifier (e.g., Figure 2 Amplifier 250) and EMI filters (e.g., Figure 2 EMI filter 260) and / or transmitter (e.g., Figure 2 (e.g., transmitter 270). Furthermore, all or a subset of any one or more of blocks 781-792 of method 780 may be determined according to the above discussion (e.g., see [link to discussion]). Figures 1-6 And execute.

[0144] Method 780 begins at block 781 by receiving signal content. In some embodiments, the signal content may be an audio signal having characteristics similar to those previously described, such as an audio bandwidth of about 60 Hz to 20 kHz and a sampling rate of 44.1 kHz or 48 kHz. The signal content may originate from various sources, such as storage devices or streaming devices.

[0145] At block 782, method 780 continues by enhancing the signal content. In some embodiments, enhancement may involve audio processing techniques such as bass boost, equalization (EQ), or other operations to improve the quality or characteristics of the signal content. The specific enhancement techniques applied may vary depending on the desired output characteristics and nature of the input signal.

[0146] At block 783, method 780 continues by upsampling the enhanced signal content. In some embodiments, upsampling may increase the sampling rate to at least 128 kHz, which can enable more accurate current estimation in subsequent processing stages. For example, the sampling rate may be quadrupled from 48 kHz to 192 kHz. Alternatively or additionally, upsampling may ensure that the maximum gradient of the upsampled signal is greater than or equal to the maximum gradient of the original enhanced signal.

[0147] At block 784, method 780 continues by limiting the oversampled content to an effective dynamic range to generate the source signal. In some cases, limiting the oversampled content to an effective dynamic range may be based on system parameters such as voltage limits or full voltage. This step helps resolve overvoltage conditions while maintaining the fidelity of lower amplitude signal content.

[0148] At block 785, method 780 continues by determining the difference between a sample of the source signal and a previous sample of the source signal. This difference calculation can be critical for estimating the possible current draw in a capacitive load, since the current in such a load is proportional to the rate of change of voltage. In some embodiments, the difference between a sample of the source signal and a previous sample of the source signal includes (a) the difference between a sample of the source signal x output from the saturation control block of the preprocessing module of the digital signal processing unit and (b) the difference between a previous sample of the source signal x output from the self-saturation control block. In these and other embodiments, the difference between a sample of the source signal and a previous sample of the source signal includes (a) the difference between a sample of the source signal x output from the self-saturation control block and (b) the difference between: (i) a filtered signal yf output from the dynamic adjustment filter of the signal conditioning module of the digital signal processing unit and corresponding to a previous sample of the source signal x, or (ii) a conditioned signal y output from the differential control block of the signal conditioning module and corresponding to a previous sample. In these and other embodiments, the difference between a sample of the source signal and a previous sample of the source signal includes (a) the difference between a sample of the filtered signal yf of the source signal x, which is output by the dynamic adjustment filter and corresponds to the output of the self-saturation control block, and (b) the difference between (i) the filtered signal yf, which is output by the dynamic adjustment filter and corresponds to the previous sample of the source signal x, or (ii) the adjusted signal y, which is output by the differential control block and corresponds to the previous sample.

[0149] At block 786, method 780 continues by determining the difference constraint (e.g., dx_lmt) based on (i) system parameters and (ii) representative values. As discussed above, the representative value xr can be the instantaneous value of the current sample of the source signal x. The maximum value between the differential threshold and a representative value yr of the regulated signal y. In some embodiments, the differential threshold may be based on both current and power constraints, thereby selecting more restrictive aspects of both. The determination may involve calculations using system parameters such as current limits, power limits, load capacitance, and sampling rate. Alternatively, the determination may involve calculations using characteristics of the source signal x and / or the regulated signal y. The differential limit is also referred to herein as a “threshold,” “differential threshold,” “differential limit threshold,” and the like.

[0150] At block 787, method 780 continues by determining the ratio of the difference limit (determined at block 786) to the difference (calculated at block 785). This ratio can be used to determine the desired gain at the Nyquist frequency, and it can also be used to determine the filter coefficients of the dynamically adjusted filter of the signal conditioning module of the digital signal processing unit of the signal driver. In some embodiments, this ratio can represent the desired level of signal conditioning to prevent excessive current draw from capacitive loads.

[0151] At block 788, method 780 continues by filtering the source signal. In some cases, a dynamically adjusted filter can be used to perform the filtering, which can be implemented as a first-order finite impulse response (FIR) filter or a first-order infinite impulse response (IIR) filter. As discussed above, the filtering can be performed at least in part based on (a) the dynamically adjusted filter coefficients determined at block 787 and (b) other system parameters.

[0152] At block 789, method 780 continues by limiting the inter-sample difference between consecutive (e.g., successive, adjacent, immediately adjacent) samples of the regulated signal y. Limiting the difference between consecutive samples of the regulated signal according to the difference limit may include generating a dynamic regulation signal. In some embodiments, limiting the inter-sample difference between consecutive samples of the regulated signal y may include limiting the magnitude of the current sample of the filtered signal yf relative to a corresponding previous sample of the regulated signal y and according to the difference limit dx_lmt. For example, limiting the inter-sample difference may include: (i) determining (a) the difference df between (a) the current sample of the filtered signal yf output by the dynamic regulation filter and (b) a previous sample (e.g., immediately adjacent previous sample) of the regulated signal y output from the difference control block. Limiting the inter-sample difference may further include comparing the magnitude d of the determined difference df with the corresponding difference limit dx_lmt.

[0153] When the magnitude of the determined difference d is less than (or equal to) the corresponding difference limit dx_lmt, the restricted inter-sample difference at block 789 may include allowing the current sample of the filtered signal yf to pass. For example, when the magnitude of the determined difference df is less than (or equal to) the corresponding difference limit dx_lmt, the determined difference df, restricted by the difference limit dx_lmt, may include allowing the gain g to pass the current sample of the filtered signal yf. As another example, when the magnitude of the determined difference df is less than (or equal to) the corresponding difference limit dx_lmt, the determined difference df, restricted by the corresponding difference limit dx_lmt, may include outputting the current sample of the adjusted signal y having a value equivalent to: (a) the current sample of the filtered signal yf, (b) the previous sample of the adjusted signal y plus the determined difference df, and / or (c) the previous sample of the adjusted signal y plus the product of the magnitude d of the determined difference df and the sign s of the determined difference df. As a specific embodiment, the restricted inter-sample difference at block 789 may include (i) determining the difference df between the current sample yf[i] of the filtered signal yf and the previous sample y[i-1] of the adjusted signal y, (ii) determining the magnitude d and sign s of the difference df, (iii) comparing the magnitude d with the difference limit dx_lmt, (iv) determining that the magnitude d is less than (or equal to) the corresponding difference limit dx_lmt and / or (v) outputting the current sample y[i] of the adjusted signal y, whose value is equivalent to the previous sample y[i-1] of the adjusted signal y plus the product of the magnitude d and the sign s.

[0154] On the other hand, when the magnitude of the determined difference df is greater than (or equal to) the corresponding difference limit dx_lmt, limiting the inter-sample difference may include limiting the magnitude of the filtered signal yf such that the magnitude of the difference between the previous sample of the adjusted signal y and the current sample of the adjusted signal y is equivalent to the corresponding difference limit dx_lmt. For example, when the magnitude of the determined difference df is greater than the corresponding difference limit dx_lmt, limiting the determined difference df according to the difference limit dx_lmt may include allowing the current sample of the filtered signal yf to pass with a gain g equivalent to a ratio of the corresponding difference limit dx_lmt to the difference df, or dx_lmt / df. The gain g may be the desired gain at the Nyquist frequency. As another example, when the magnitude d of the determined difference df is greater than (or equal to) the corresponding difference limit dx_lmt, the determined difference df, according to the corresponding difference limit dx_lmt, may include the current sample of the regulated signal y with the output having a value equivalent to the following: the previous sample of the regulated signal y plus the product of the corresponding difference limit dx_lmt and the sign s of the determined difference df. As a specific embodiment, the restricted sample difference at block 789 may include (i) determining the difference df between the current sample yf[i] of the filtered signal yf and the previous sample y[i-1] of the adjusted signal y, (ii) determining the magnitude d and sign s of the difference df, (iii) comparing the magnitude d with the difference limit dx_lmt, (iv) determining that the magnitude d is greater than the corresponding difference limit dx_lmt, (v) limiting the magnitude d by setting the magnitude d to be equal to the difference limit dx_lmt, and / or (vi) outputting the current sample y[i] of the adjusted signal y, whose value is equivalent to the previous sample y[i-1] of the adjusted signal y plus the product of the limited magnitude d and the sign s.

[0155] At block 790, method 780 continues by generating an amplified signal based on a dynamic adjustment signal. In some embodiments, this amplification may be performed by a Class D or Class AB amplifier having specified rated voltage, current, and power.

[0156] At block 791, method 780 continues by filtering the amplified signal. In some embodiments, filtering may be performed by an electromagnetic interference (EMI) filter, which may include an LC filter or a ferrite bead filter designed to reduce unwanted frequencies or noise.

[0157] At block 792, method 780 terminates by transmitting an output signal. In some embodiments, the output signal may be transmitted by a capacitive load such as a piezoelectric speaker.

[0158] Although blocks 781-792 of method 780 are discussed and shown in a specific order, Figure 7Method 780 is not limited thereto. In other embodiments, all or a subset of one or more of blocks 781-792 of method 780 may be executed in a different order. In these and other embodiments, all or a subset of any of blocks 781-792 of method 780 may be executed before, during, and / or after all or a subset of any of the other blocks 781-792 of method 780. Furthermore, those skilled in the art will readily recognize that method 780 may be modified while still remaining within these and other embodiments of the present invention. For example, all or a subset of one or more blocks 781-792 of method 780 may be omitted and / or repeated in some embodiments. As a specific example, in an embodiment omitting the EMI filter, block 791 may be omitted from method 780.

[0159] 5. Representative Results / Simulation

[0160] Figure 8 This set of three vertically aligned curves 891, 892, and 893 illustrates different states of signal processing for capacitive loads according to various embodiments of the present invention. Curves 891-893 share a common x-axis representing time in seconds.

[0161] Curve 891 illustrates the amplitude of the audio signal generated using the glockenspiel over time. More specifically, curve 891 shows the amplitude of the original audio signal sampled at 48 kHz, the oversampled audio signal resampled at 196 kHz, the compressed / filtered signal corresponding to the oversampled audio signal (shown in dark gray), and the adjusted signal corresponding to the compressed / filtered signal (shown in light gray). The y-axis of curve 891, ranging from -1 to 1, represents the normalized amplitude of the signal. Curve 891 illustrates how the original audio signal is processed and modified at different stages of the adaptive current-limiting process according to various embodiments of the invention.

[0162] Curve 892 illustrates the changes in the saturation gain G and the filter coefficient B1 of the finite impulse response (FIR) filter used over time when processing the audio signal of curve 891 according to various embodiments of the invention. The y-axis of curve 892 is in the range of 0 to 1, within which the saturation gain and filter coefficient values ​​fluctuate. Curve 892 shows multiple different time periods in which the value of the filter coefficient B1 changes rapidly, thereby indicating the dynamic adjustments in signal processing to limit current according to the above discussion of the invention.

[0163] Curve 893 depicts the change over time of the differential limit applied to the upsampled audio signal by the signal conditioning module configured according to various embodiments of the present invention. The y-axis of curve 893, ranging from 0 to 0.1, represents the magnitude of the differential limit. This curve illustrates the rapid fluctuations in the differential limit, corresponding to the dynamic nature of the signal processing algorithm. As discussed above, the differential limit can be adjusted based on various factors such as signal amplitude, system parameters, and current or power constraints.

[0164] Curves 891-893 illustrate how the revealed method can dynamically adjust signal processing parameters to maintain signal fidelity while keeping current consumption within desired limits. The adaptive nature of the revealed technique is expected to enable more flexible and efficient signal conditioning, especially compared to fixed filtering methods.

[0165] More specifically, Figure 9 This section compares two spectrograms, represented by curves 997 and 999, illustrating two different signal processing methods applied to the same audio signal. In both graphs, the x-axis represents time from 0.0 to 7.0 seconds, while the y-axis represents frequency from 0 to 24 kHz. The spectrograms use grayscale intensity to represent signal strength at different frequencies over time.

[0166] Curve 997, labeled “Adaptive Current Limiter,” illustrates the spectrogram of an audio signal processed using adaptive current limiting technology and / or dynamically adjusted filters according to various embodiments of the invention. As discussed in more detail below, curve 997 shows how the adaptive current limiter preserves more frequency content of the audio signal across time, specifically at higher frequencies.

[0167] Curve 999, labeled "Fixed 10 kHz Bandwidth Limiter," displays the same audio signal but processed using a fixed bandwidth limiter, which can represent an existing low-pass filter with a fixed bandwidth. Curve 999 shows a more uniform attenuation of frequencies above 10 kHz across all time periods.

[0168] The comparison between curves 997 and 999 illustrates the advantages of the adaptive current limiting technology of this invention compared to a fixed bandwidth limiter. Specifically, as shown in curve 997, the adaptive current limiter maintains higher frequency audio signal content where possible, resulting in better audio quality. In contrast, as shown in curve 999, the fixed bandwidth limiter continuously attenuates frequencies above 10 kHz, leading to a significant loss of audio quality.

[0169] C. Conclusion

[0170] The detailed description of embodiments of the present invention above is not intended to be exhaustive or to limit the technology to the precise forms disclosed herein. While specific embodiments and examples of the technology have been described above for illustrative purposes, those skilled in the art will understand that various considerable modifications can be made within the scope of this technology. For example, although the steps are presented in the order given above, alternative embodiments may perform the steps in a different order. Furthermore, the various embodiments described herein may be combined to provide other embodiments.

[0171] Based on the foregoing, it should be understood that specific embodiments of the technology have been described herein for illustrative purposes, but well-known structures and functions have not been shown or described in detail to avoid unnecessarily obscuring the description of embodiments of the technology.

[0172] Where the context permits, singular or plural terms may also include plural or singular terms, respectively. Furthermore, unless the word “or” is explicitly limited to a single item that is exclusive to other items in a list referring to two or more items, its use in this list may be interpreted to include: (a) any single item in the list, (b) all items in the list, or (c) any combination of items in the list. Additionally, as used herein, the phrase “and / or” in phrases such as “A and / or B” means only A, only B, and both A and B. Furthermore, the terms “comprising,” “including,” “having,” and “with” are used throughout to mean that at least one or more of the described features are included, such that no larger number of identical features and / or other features of additional types are excluded. Furthermore, as used herein, the phrases “based on,” “depending on,” “due to,” and “in response to” should not be construed as references to a closed set of conditions. For example, an exemplary step described as “based on condition A” may be based on both condition A and condition B without departing from the scope of this disclosure. In other words, as used in this article, the phrase “based on” should be interpreted in the same way as the phrases “based at least in part on” or “based atleast partially on”.

[0173] Based on the foregoing, it should also be understood that various modifications can be made without departing from this disclosure or the technology of the present invention. For example, those skilled in the art will understand that the various components of the present invention can be further divided into sub-components, or the various components and functions of the present invention can be combined and integrated. Furthermore, in other embodiments, certain aspects of the technology described in the context of a particular embodiment can be combined or eliminated. Moreover, although advantages associated with other embodiments of the present technology have been described in the context of some embodiments, other embodiments may also exhibit such advantages, and not all embodiments necessarily need to exhibit such advantages to fall within the scope of the present technology. Therefore, this disclosure and related technologies may cover other embodiments not explicitly shown or described herein.

Claims

1. A signal transmission system, characterized in that, Include: The signal processing unit is configured as follows: The difference constraint is determined at least in part based on representative values ​​corresponding to multiple system parameters and samples of the source signal, one or more samples of the regulated signal output by the signal processing unit, or combinations thereof, wherein the difference constraint represents a constraint on the first difference between consecutive samples of the regulated signal. Determine the ratio of the difference limit to the second difference between the sample of the source signal and a previous sample of the source signal, and A low-pass filter is applied to the source signal to obtain a filtered signal, wherein the coefficients of the low-pass filter are generated at least in part based on the ratio; An amplifier coupled to the output of the signal processing unit and configured to generate an amplified signal based at least in part on the regulated signal; as well as A transmitter configured to transmit an output signal based at least in part on the amplified signal.

2. The signal transmitting system according to claim 1, characterized in that, The signal processing unit is further configured to limit a third difference between a sample of the filtered signal and a previous sample of the regulated signal based on the difference constraint.

3. The signal transmitting system according to claim 2, characterized in that, When the value of the third difference is greater than the difference limit, the signal processing unit is configured to generate a sample of the regulated signal such that the value of the fourth difference between the sample of the regulated signal and the previous sample of the regulated signal is less than or equal to the difference limit.

4. The signal transmitting system according to claim 2, characterized in that, When the magnitude of the third difference is greater than the difference limit, the signal processing unit is configured to generate a sample of the regulated signal, the sample having a value equivalent to the sum of the following: the previous sample of the regulated signal and the product of the difference limit and the sign of the third difference.

5. The signal transmitting system according to claim 2, characterized in that, When the value of the third difference is less than or equal to the difference limit, the signal processing unit is configured to generate a sample of the regulated signal, the sample having a value equivalent to the sum of the following: the previous sample of the regulated signal and the third difference.

6. The signal transmitting system according to claim 2, characterized in that, This representative value is equivalent to the maximum value between the instantaneous value of the source signal and the representative value of the adjusted signal.

7. The signal transmitting system according to claim 6, characterized in that, This representative value of the adjusted signal corresponds to the maximum of the following two: The instantaneous value of the current sample of the adjusted signal; and The magnitude of a function obtained from one or more previous samples based on the regulated signal.

8. The signal transmitting system according to claim 1, characterized in that, The difference is limited to the minimum between the first and second values, where the first value is equivalent to dx_i = Ix / (Fs * C * Vf), and the second value is equivalent to dx_p = ((Px / (Fs * C * Vf)). 2 )) / xr, where Ix is the current limit, Fs is the sampling rate, C is the load capacitance, Vf is the full voltage, Px is the power limit, and xr is the representative value.

9. The signal transmitting system according to claim 1, characterized in that, The multiple system parameters include: Full voltage Vf; Voltage limit Vx; The current limit Ix represents the minimum of the amplifier's rated current, the rated current of any component connected in series with the transmitter, and the saturation current of any component connected in series with the transmitter. Power limit Px, which represents the minimum rated power of the amplifier and the rated power of the transmitter; Sampling rate Fs; and Load capacitance, which represents the capacitance of the transmitter, or the combined capacitance of the transmitter and the LC filter coupled between the signal processing unit and the transmitter.

10. The signal transmitting system according to claim 1, characterized in that, This low-pass filter includes a finite impulse response filter.

11. The signal transmitting system according to claim 10, characterized in that, The signal processing unit is further configured to dynamically determine the coefficients of the finite impulse response filter according to the following equation: B0 = (1+g) / 2, and B1 = (1-g) / 2; Where B0 and B1 are the coefficients of the finite impulse response filter, and g is the ratio.

12. The signal transmitting system according to claim 1, characterized in that, This low-pass filter includes an infinite impulse response filter.

13. The signal transmitting system according to claim 12, characterized in that, The signal processing unit is further configured to dynamically determine the coefficients of the infinite impulse response filter according to the following equation: B0 = 2g / (1+g), and A1 = (1-g) / (1+g); Where B0 and A1 are the coefficients of the infinite impulse response filter, and g is the ratio.

14. The signal transmitting system according to claim 1, characterized in that, The source signal is an audio signal with a sampling rate of at least 128 kHz.

15. The signal transmitting system according to claim 1, characterized in that, The transmitter includes a piezoelectric speaker.

16. The signal transmitting system according to claim 1, characterized in that: The amplifier is a Class D amplifier or a Class AB amplifier; or The signal transmission system further includes an electromagnetic interference filter coupled between the amplifier and the transmitter.

17. The signal transmitting system according to claim 1, characterized in that, The signal processing unit is further configured to limit the voltage of the source signal within an effective dynamic range, which is based at least in part on system parameters including full voltage and voltage limit; and wherein the voltage limit represents the minimum value of the supply voltage of the amplifier, the rated voltage of the capacitive load of the transmitter, and / or the rated voltage of the capacitor of the LC filter coupled between the signal processing unit and the transmitter.

18. A signal conditioning module, characterized in that, For use in signal transmission systems, the signal conditioning module includes: A dynamically adjustable filter configured to filter a source signal at least partially based on filter coefficients to produce a filtered signal; and A constraint / coefficient generator, configured as follows: The difference limit is determined at least in part based on a number of system parameters and representative values ​​corresponding to samples of the source signal, one or more samples of the regulated signal output by the signal conditioning module, or a combination thereof, wherein the difference limit represents a limit on the first difference between consecutive samples of the regulated signal; Determine the ratio of the difference limit to the second difference between the sample of the source signal and a previous sample of the source signal; and Multiple filter coefficients are generated, at least in part, based on this ratio.

19. The signal conditioning module according to claim 18, characterized in that, It further includes a differential control block configured to limit the first difference by limiting a third difference between a sample of the filtered signal and a previous sample of the regulated signal according to the differential limit.

20. A signal conditioning method, characterized in that, The source signal is processed into a regulated signal for transmission as an amplified signal via a capacitive load. This signal regulation method includes: The difference limit is determined at least in part based on multiple system parameters and representative values ​​corresponding to samples of the source signal, one or more samples of the regulated signal, or combinations thereof, wherein the difference limit represents a limit on the first difference between two samples of the regulated signal; Determine the ratio of the difference limit to the difference between the current sample of the source signal and a previous sample of the source signal; as well as A low-pass filter is applied to the source signal to obtain a filtered signal, wherein the coefficients of the low-pass filter are generated at least in part based on the ratio; The regulated signal is generated at least in part based on the filtered signal; and An amplified signal is generated, at least in part, based on the regulated signal, for transmission via the capacitive load.

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