A high-pass and low-pass filter co-tunable DSP audio filter system and method

Abstract

The application relates to the technical field of digital signal processing audio systems, in particular to a DSP audio filter system and method with high-pass and low-pass filters capable of being jointly adjusted, which comprises an analog-digital conversion module, a digital-analog conversion module, a digital signal processing module used for filtering and processing a digital audio signal, a user control interface, and a joint adjustment control module which pairs at least one high-pass filter with at least one low-pass filter in response to a joint adjustment mode selected by a user through the user control interface; when the user adjusts the cut-off frequency point of any filter in the pair in the joint adjustment mode, the joint adjustment control module controls the cut-off frequency point of the other filter in the pair to change synchronously and equivalently. The beneficial effect lies in that, through the joint adjustment of the high-pass and low-pass filters, the user only needs to adjust one parameter to synchronously change the cut-off frequency points of the paired filters, so that the operation process of the frequency division point setting is greatly simplified, and the tuning efficiency is improved.

Description

Technical Field

[0001] This invention relates to the field of digital signal processing audio system technology, and in particular to a DSP audio filter system and method with adjustable high-pass and low-pass filters. Background Technology

[0002] In modern digital audio systems, digital signal processors (DSPs) are typically used to cross-process audio signals to achieve separate outputs for different frequency bands. Common filter types include Butterworth, Bessel, and Linkwitz-Riley high-pass, low-pass, and band-pass filters. In traditional DSP crossover systems, each filter parameter needs to be set independently. For example, in a two-way system, the user needs to set the cutoff frequencies (cut-off points) of the high-pass and low-pass filters separately. This process is cumbersome and can easily lead to mismatched frequency response curves due to asynchronous settings, affecting the listening experience.

[0003] In addition, existing systems typically lack the ability to quickly and smoothly switch between crossover mode and full-frequency mode, making it inconvenient for users to switch between different listening scenarios.

[0004] Therefore, it is necessary to provide a DSP filter system that is easier to operate, has a more coordinated response, and supports flexible switching between frequency division and full frequency.

[0005] Application content

[0006] The present invention proposes a DSP audio filter system and method with adjustable high-pass and low-pass filters to solve the above problems.

[0007] The technical solution of this invention is implemented as follows:

[0008] A DSP audio filter system with adjustable high-pass and low-pass filters, comprising:

[0009] An analog-to-digital converter module is used to convert input analog audio signals into digital signals;

[0010] A digital signal processing module, which includes at least one high-pass filter and at least one low-pass filter for filtering digital audio signals;

[0011] The digital-to-analog converter module is used to convert the processed digital signal into an analog signal for output.

[0012] The user control interface is used to receive user operation commands.

[0013] The joint debugging control module, in response to the joint debugging mode selected by the user through the user control interface, pairs and links at least one high-pass filter with at least one low-pass filter.

[0014] In the joint debugging mode, when the user adjusts the frequency cut point of any filter in the pair, the joint debugging control module controls the frequency cut point of the other filter in the pair to change synchronously and equally.

[0015] Furthermore, the frequency cut-off point adjustment range of the high-pass filter and the low-pass filter is the effective audio frequency range of the system, preferably 20Hz to 20000Hz. When the frequency cut-off point of the paired high-pass and low-pass filters is adjusted to one extreme value of the range, the system realizes the conversion between frequency division mode and full-frequency mode.

[0016] Furthermore, the joint debugging control module also supports full joint debugging functions. When the system has multiple identical audio processing channels, after the user completes the joint debugging settings on one channel, the same settings can be copied to other channels through the full joint debugging function.

[0017] Furthermore, the user control interface is at least one of a physical knob / button panel, a touch screen, a mobile application, or a computer software interface.

[0018] Furthermore, the types of filters in the digital signal processing module include Butterworth, Bessel, or Linkowitz filters, and the co-tuning function is applicable to all supported filter types.

[0019] This invention also provides a DSP audio filter co-tuning control method based on the above-mentioned DSP audio filter system with adjustable high-pass and low-pass filters, comprising the following steps:

[0020] S1 system initialization, loading default or user-saved filter parameters;

[0021] S2 receives user input, allowing users to select between joint debugging mode and monotonic mode;

[0022] If the S3 selects the joint debugging mode, it receives the pairing relationship between the high-pass filter and the low-pass filter specified by the user.

[0023] S4 receives user commands to adjust the frequency switching point of any filter in the pair;

[0024] S5 adjusts the cutoff frequency of the other filter in the pair synchronously and equally according to the adjustment command;

[0025] S6 outputs the filtered signal after linkage adjustment in real time.

[0026] Furthermore, it also includes a frequency division / full-range mode switching step: when the cut-off points of the paired high-pass and low-pass filters are synchronously adjusted to the upper limit of the audio frequency range, the high-pass filter has no signal output, the low-pass filter outputs a full-range signal, and the system is in full-range mode; when the cut-off points are synchronously adjusted to the lower limit of the audio frequency range, the low-pass filter has no signal output, the high-pass filter outputs a full-range signal, and the system is also in full-range mode; when the cut-off points are between the upper and lower limits, the system is in frequency division mode.

[0027] By adopting the above technical solution, the beneficial effects of the present invention are as follows:

[0028] 1. By adjusting the high-pass and low-pass filters in tandem, users only need to adjust one parameter to synchronously change the crossover point of the paired filter, which greatly simplifies the operation process of setting the crossover point and improves the tuning efficiency.

[0029] 2. The joint tuning mode ensures strict synchronization and equal value change of the paired filter's frequency switching point, avoiding possible errors from manual adjustment, making the connection between the high-pass and low-pass frequency bands more accurate and smooth, and optimizing the frequency response curve.

[0030] 3. By utilizing the joint debugging mechanism, the frequency switching point can be adjusted to both ends of the frequency range, allowing seamless and linear switching between frequency division mode and full-frequency mode, thus enhancing the system's adaptability and application flexibility.

[0031] 4. The joint debugging function is implemented by software algorithm without changing the hardware circuit structure. It can be easily integrated into existing DSP audio processing platforms or power amplifier equipment. All joint debugging functions further simplify the consistency settings of multi-channel systems.

[0032] 5. The joint debugging technology is applicable to various types of digital filters and frequency divider, frequency divider, and even multi-frequency divider system architectures, and has broad application prospects. Attached Figure Description

[0033] To more clearly illustrate the technical solutions in the embodiments of the present invention or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present invention. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0034] Figure 1 This is a system structure block diagram of the present invention;

[0035] Figure 2 This is a schematic diagram showing the synchronous change of the switching points of the high-pass and low-pass filters under the joint debugging mode of this invention;

[0036] Figure 3This is a schematic diagram illustrating the principle of frequency division and full-frequency mode conversion achieved through joint debugging in this invention;

[0037] Figure 4 This is an example diagram of the joint debugging mode setting page in the user control interface of this invention. Detailed Implementation

[0038] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0039] like Figure 1 As shown, a DSP audio filter system with adjustable high-pass and low-pass filters includes:

[0040] The analog-to-digital converter (ADC) is responsible for receiving analog audio signals from an audio source (such as a host or player) and converting them into a digital audio signal stream at a predetermined sampling rate (such as 48kHz) and quantization precision.

[0041] The digital signal processing module (DSP Core) has embedded digital filtering algorithm programs. It can generate one or more digital processing instances of high-pass filters (HPF), low-pass filters (LPF), and band-pass filters (BPF) composed of the two in real time according to the configuration parameters. Each filter instance contains adjustable parameters such as type (Butterworth, Bessel, etc.), cut-off frequency, and slope, which are stored in the corresponding parameter register.

[0042] The Linkage Control Unit runs on the DSP core or a microcontroller tightly coupled to it. It maintains a "linkage relationship table" to record which HPF instances and which LPF instances are bound as linkage pairs. When it detects that the user has adjusted the frequency cutting point of any filter in the linkage pair, the module immediately sends an instruction to the DSP core to update the frequency cutting point parameter register of the paired filter to the same value.

[0043] The user control interface, serving as the entry point for human-computer interaction, can be an in-vehicle physical panel, a touch screen, or computer software or a mobile APP connected via USB / Wi-Fi. This interface provides a graphical interface, including controls such as "mode selection" (monotonic / coupled tuning), "filter selection", and "parameter slider".

[0044] The digital-to-analog converter (DAC) converts the digital audio stream processed by the DSP back into an analog signal and outputs it to the subsequent power amplifier.

[0045] Combination Figure 2 The software implementation process of the "integration debugging" function of this invention is as follows:

[0046] Step S1: Mode Selection and Pairing. The user selects "Integration Mode" through the interface. Then, in the filter configuration interface, the user selects an HPF (e.g., CH1) by checking or dragging. HPF ) with an LPF (e.g., CH2) LPF This is designated as a linkage pair. This operation essentially creates a record in the "Linkage Relationship Table".

[0047] Step S2: Parameter synchronization adjustment. When the user adjusts the switching frequency point of the HPF in the linkage alignment from f using the slider or knob... old Change to f new At that time, the user interface sends a parameter update command.

[0048] Step S3: The joint debugging control module responds. Upon intercepting the instruction, the joint debugging control module first queries the "Joint Debugging Relationship Table" to confirm CH1. HPF With CH2 LPF There is a binding relationship.

[0049] Step S4: Parameter synchronous writing. The module then sends two parameter write commands to the DSP core: one is to write CH1... HPF The switching frequency is set to f. new Secondly, CH2 is simultaneously LPF The switching frequency is also set to f. new .

[0050] Step S5: Real-time processing. The DSP core immediately uses the updated parameters to perform filtering operations. For the input signal, values ​​higher than f... new The main components are output from the HPF channel, below f new The main component is output from the LPF channel. Since the two cutting frequencies are strictly equal and change synchronously, its frequency response curve is at f new This achieves precise alignment, avoiding frequency gaps or overlaps that may occur due to manual adjustments.

[0051] Figure 3 The process of achieving mode conversion through joint debugging is clearly demonstrated.

[0052] Frequency division mode: such as Figure 3 As shown in (a), when the linkage frequency switching point f cWhen set within the effective audio range (20Hz-20kHz) (e.g., 2000Hz), the system operates in a typical frequency division mode, with the HPF and LPF each performing their respective functions, processing high-frequency and low-frequency signals respectively.

[0053] Switching to full-frequency mode: When the user switches the frequency point f during joint debugging c Continuously adjust towards higher frequencies.

[0054] When f c It was adjusted to near and eventually equal to the upper limit frequency f allowed by the system. high (e.g., at 20kHz) such as Figure 3 As shown in (b), the cutoff frequency of the HPF is higher than the highest audio component, so no actual signal passes through its passband and the output is almost zero; while the cutoff frequency of the LPF is 20kHz, and its passband covers the entire audio range (20Hz-20kHz), so all signals are actually output from the LPF channel, and the system is equivalent to a full-frequency system.

[0055] Reverse conversion: Conversely, if f c Continuously adjust to the lower end of the frequency range until the lower limit f is reached. low (For example, 20Hz), the LPF passband narrows to almost no output, while the HPF passband covers the entire frequency range, and the system is equivalent to outputting a full-frequency signal from the HPF channel.

[0056] This function is not simply a bypass filter, but rather, through extreme tuning parameters, it dynamically degrades one filter into a "transparent channel" and transforms the other into a "full-pass filter" by utilizing the inherent frequency response characteristics of the filters. The entire process is continuous and linear, and the user can achieve mode transitions by smoothly moving a single slider without switching hardware circuits or changing the system topology, demonstrating the great flexibility of software control.

[0057] Application scenarios:

[0058] Stereo Systems and "Full Integration": In car stereo systems, the left and right channels typically need to be set symmetrically. Users should complete the setup for the left channel (CH1). HPF With CH2 LPF After configuring the joint debugging settings, the "All Joint Debugging" function can be enabled. At this time, the joint debugging control module will perform parameter copying and relationship copying: not only copying the right channel (CH3)... HPF With CH4 LPF The frequency cut point of the right channel is copied to the same value, and the same HPF-LPF binding relationship is established for the right channel in the "Integration Relationship Table". This ensures the rapid unification of the multi-channel system configuration.

[0059] Each joint debugging operation operates on the same internal mechanism as the core of this invention, involving synchronous and equal-value adjustment between HPF and LPF (or equivalent LPF). This demonstrates the universality of this invention.

[0060] Components not described in detail in this article are existing technologies.

[0061] The above description is only a preferred embodiment of the present invention and is not intended to limit the present invention. Any modifications, equivalent substitutions, improvements, etc., made within the spirit and principles of the present invention should be included within the protection scope of the present invention.

Claims

1. A DSP audio filter system with adjustable high-pass and low-pass filters, characterized in that: include: An analog-to-digital converter module is used to convert input analog audio signals into digital signals; A digital signal processing module, which includes at least one high-pass filter and at least one low-pass filter for filtering digital audio signals; The digital-to-analog converter module is used to convert the processed digital signal into an analog signal for output. The user control interface is used to receive user operation commands. The joint debugging control module, in response to the joint debugging mode selected by the user through the user control interface, pairs and links at least one high-pass filter with at least one low-pass filter. In the joint debugging mode, when the user adjusts the frequency cut point of any filter in the pair, the joint debugging control module controls the frequency cut point of the other filter in the pair to change synchronously and equally.

2. The DSP audio filter system with adjustable high-pass and low-pass filters according to claim 1, characterized in that: The frequency cut-off point adjustment range of the high-pass filter and the low-pass filter is the effective audio frequency range of the system, preferably 20Hz to 20000Hz. When the frequency cut-off point of the paired high-pass and low-pass filters is adjusted to one extreme value of the range, the system realizes the conversion between frequency division mode and full-frequency mode.

3. The DSP audio filter system with adjustable high-pass and low-pass filters according to claim 1, characterized in that: The joint debugging control module also supports all joint debugging functions. When the system has multiple identical audio processing channels, after the user completes the joint debugging settings on one channel, the same settings can be copied to other channels through the all joint debugging function.

4. The DSP audio filter system with adjustable high-pass and low-pass filters according to claim 1, characterized in that: The user control interface is at least one of a physical knob / button panel, a touch screen, a mobile application, or a computer software interface.

5. A DSP audio filter system with adjustable high-pass and low-pass filters according to claim 1, characterized in that: The digital signal processing module includes Butterworth, Bessel, or Linkowitz filters, and the integration function is applicable to all supported filter types.

6. A DSP audio filter co-tuning control method based on a DSP audio filter system with co-tunable high-pass and low-pass filters according to any one of claims 1-5, comprising the following steps: S1 system initialization, loading default or user-saved filter parameters; S2 receives user input, allowing users to select between joint debugging mode and monotonic mode; If the S3 selects the joint debugging mode, it receives the pairing relationship between the high-pass filter and the low-pass filter specified by the user. S4 receives user commands to adjust the frequency switching point of any filter in the pair; S5 adjusts the cutoff frequency of the other filter in the pair synchronously and equally according to the adjustment command; S6 outputs the filtered signal after linkage adjustment in real time.

7. The DSP audio filter co-tuning control method for high-pass and low-pass filters according to claim 6, characterized in that: It also includes a frequency division / full-range mode switching step: when the cut-off points of the paired high-pass and low-pass filters are synchronously adjusted to the upper limit of the audio frequency range, the high-pass filter has no signal output, the low-pass filter outputs a full-range signal, and the system is in full-range mode; when the cut-off point is synchronously adjusted to the lower limit of the audio frequency range, the low-pass filter has no signal output, the high-pass filter outputs a full-range signal, and the system is also in full-range mode; when the cut-off point is between the upper and lower limits, the system is in frequency division mode.

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