A method, system and audio system for automatic spatial sound field correction

Abstract

This invention relates to an automatic spatial sound field correction method and system, and an audio system, comprising the following steps: connecting an automatic equalization (EQ) application unit to a test terminal via Bluetooth and the audio system; scanning to obtain a near-field curve at the near-field position of the audio system; obtaining a far-field curve through a first scan at the far-field position, wherein the near-field and far-field curves are two synchronously formed curves based on the same frequency range as the abscissa during scanning; applying a predetermined algorithm to perform appropriate peak compression or valley filling on the far-field curve to obtain an adjusted initial EQ curve; the data signal processing unit in the audio system processes audio data based on the obtained initial EQ curve, and then, at the same far-field position, performs a second scan through the test terminal to obtain a far-field quadratic curve, thus obtaining the final frequency response curve for correction. The entire automatic equalization adjustment process can be performed on terminals such as mobile phones according to the user's personalized selection, achieving real-time dynamic adjustment and meeting the user's personalized needs, and can be widely applied to various audio equipment.

Description

Technical Field

[0001] This invention relates to the field of audio processing technology, specifically to a method, system, and audio system for automatic spatial sound field correction. Background Technology

[0002] As people's living standards continue to improve, their demand for daily entertainment is also increasing. Audio equipment has become one of the most important facilities for mass entertainment, car audio systems, and entertainment venues, especially an indispensable facility for celebrations and festivals. For the average consumer, connecting audio equipment to mobile devices such as smartphones to play music from their phones or enjoy the surround sound effect of a movie theater while watching videos has become the mainstream demand for audio equipment.

[0003] Currently, DSP (Digital Signal Processing Unit) technology is mature, offering higher precision, lower distortion, and more flexible algorithms, gradually replacing analog hardware. Traditional smart speakers and car audio systems often feature "one-click automatic tuning," however, these systems only support single calibration, lack real-time dynamic adjustment, and struggle to meet users' personalized needs. Summary of the Invention

[0004] In view of this, a method, system and audio system for automatic spatial sound field correction that can be dynamically adjusted in real time and meet the personalized needs of users are provided.

[0005] An automatic spatial sound field correction method for automatically correcting the EQ curve of a sound system within a space, characterized by comprising the following steps: The test terminal's automatic equalization EQ application unit and speakers are connected via Bluetooth. Near-field curves were obtained by scanning at the near-field position of the audio equipment using a test terminal. At a predetermined distance from the speaker, the far-field curve is obtained through the first scan using the test terminal. The far-field curve and the near-field curve are two synchronous curves formed based on the same frequency range as the horizontal axis during the scan, so that they can be superimposed in the same coordinate system for synchronous adjustment and calculation. By using a predetermined algorithm, the far-field curve is subjected to corresponding peak suppression or valley filling to obtain the adjusted EQ curve; The data signal processing unit in the audio system processes the audio data based on the obtained initial EQ curve, and then obtains the far-field quadratic curve by a second scan through the test terminal at the same far-field position, thus obtaining the final frequency response curve for correction. Each of the above scanning steps is performed in the automatic equalization (EQ) application unit of the test terminal.

[0006] Furthermore, the far-field curve and the near-field curve are generated in the same EQ curve graph. The far-field curve and the near-field curve are frequency response data curves formed at positions close to and far from the speaker based on the same test signal. The near-field curve is generated by measuring the maximum sound pressure level at each frequency point using a microphone at the near-field position of the speaker. The far-field curve is generated by measuring the maximum sound pressure level at each frequency point using a listening position at a predetermined distance from the speaker.

[0007] Furthermore, the predetermined algorithm includes setting peak attenuation filters at and near each peak of the far-field curve, such that the attenuation amplitude of the far-field curve at and near the peak is -3.5 to -1 dB, and setting Shelf filters at and near each trough of the far-field curve, such that the gain amplitude of the far-field curve at and near the trough is 3 to 6 dB, and the gain decreases linearly from the trough to both sides of the frequency band.

[0008] Furthermore, the predetermined algorithm includes setting a peak attenuation filter in the 50Hz-65Hz frequency band, so that the attenuation amplitude of the far-field curve in this frequency range is -3.5 to -1 dB, and the peak attenuation amplitude decreases linearly from 50Hz to 65Hz; setting a peak attenuation filter in the 30Hz-33Hz frequency band, so that the attenuation amplitude of the far-field curve in this frequency range is -3 to -1 dB; setting a low-frequency Shelf filter in the 33Hz-50Hz frequency band, so that the gain amplitude of the far-field curve in this frequency range is 3 to 6 dB, and the gain decreases linearly from the trough to the two adjacent frequency bands; and setting a high-frequency Shelf filter in the 100Hz-120Hz frequency band, so that the gain amplitude of the far-field curve in this frequency range is 1 to 3 dB, and the gain decreases linearly from the trough to the two adjacent frequency bands.

[0009] Furthermore, when obtaining the near-field curve, the maximum sound pressure level at each frequency point is measured using a test terminal at the near-field position of the speaker, and the data is stored in the Near[] array; when obtaining the far-field curve, the maximum sound pressure level at each frequency point is measured using a test terminal at a listening position at a predetermined distance from the speaker, and the data is stored in the Target[] array; the frequency range of the test is 20-120Hz, and the total length of each of the following arrays is 101-120Hz; the predetermined algorithm includes the following steps: (1) Standardization of benchmark data a. Calculate the average value Near_mid of the Near[] array using the following formula: Near_mid = Sum of maximum sound pressure levels in the near field / Total frequency range; b. Perform the following transformation on each value N_i in the Near[] array: N'_i = N_i - Near_mid, where i is the index of the corresponding frequency and the range of i is 0-100; (2) Standardization of target data a. Calculate the average value Target_mid of the Target[] array using the following formula: Target_mid = Sum of maximum sound pressure levels in the far field / Total frequency range; b. Perform the transformation on each value T_j in the Target[] array: T'_j = T_j - Target_mid, where j is the index of the corresponding frequency and the range of j is 0-100; (3) Shape difference analysis Based on the Near[] array and the Target[] array, the standardized spectral difference is calculated as follows: Diff_k = T'_k - N'_k, resulting in the Diff[] array, where k is the index of the corresponding frequency, ranging from 0 to 100, T'_k is the converted value T'_j of the Target[] array at the corresponding frequency, and N'_k is the converted value N'_i of the Near[] array at the corresponding frequency. (4) Key feature point extraction Scan the Diff[] array to extract feature points, and store the results in the Peak[] array. The feature points include peaks and troughs. (5) EQ parameter generation a. Select the M feature points with the largest amplitude in the Peak[] array, where M ranges from 6 to 12; b. Calculation rules: Obtain the starting index of Peak, denoted as Left, Peak, Right; Calculate the bandwidth (BW) and quality factor (Q) of the Peak. The bandwidth (BW) of the Peak is defined as follows: for a center frequency Fc, there exists a left-hand extreme frequency and a right-hand extreme frequency. The spectral difference (Diff_k) between the left-hand extreme frequency and the right-hand extreme frequency is close to 0. The bandwidth (BW) is equal to the absolute value of the difference between the left-hand and right-hand extreme frequencies. The formula for calculating the quality factor (Q) is as follows: Q = Fc / BW; G = (Diff[Left]+Diff[Right]) / 2 - Diff[Peak], where Diff[Left] is the spectral difference of the frequency points to the left of the peak or valley, Diff[Right] is the spectral difference of the frequency points to the right of the peak or valley, and Diff[Peak] is the spectral difference of the current peak or valley point; (6) Output: Output M sets of EQ parameters, each set of EQ parameters includes center frequency Fc, gain value G, and Q value.

[0010] Furthermore, obtaining the final frequency response curve for correction by testing the far-field quadratic curve a second time using the test terminal specifically includes the following steps: The obtained initial EQ curve is input into the data signal processing unit; wherein, the initial EQ curve or EQ parameter data is first obtained by communicating with the test terminal through a main control chip, and then transmitted to the transceiver data signal processing unit by the main control chip; Based on the obtained initial EQ curve, the data signal processing unit processes the audio data of the frequency band within the band BW of each center frequency according to a corresponding set of EQ parameters and plays it. Then, at the same far-field location, the frequency response curve is obtained by measuring with a test terminal, resulting in the final frequency response curve used for calibration.

[0011] Further, the step of applying a predetermined algorithm to suppress peaks or fill valleys in the far-field curve includes constructing an attenuation gain curve according to the predetermined algorithm, and then superimposing the attenuation gain curve with the far-field curve to obtain the adjusted initial EQ curve; the automatic equalization EQ application unit is equipped with a phase adjustment unit, a delay adjustment unit, a polarity adjustment unit, a filter type selection unit, an equalization preset unit, an automatic equalization operation unit, and an automatic equalization interface unit. Each unit is used by the user to perform personalized equalization settings on the terminal. The phase adjustment unit is used for phase adjustment, with a phase adjustment range of 0º-180º; the delay adjustment unit is used for delay adjustment, with a delay adjustment range of 0-20ms; the polarity adjustment unit is used for positive and negative polarity adjustment to switch between positive and negative polarities, changing from 0º to 180º when turned on; the filtering... The filter type selection unit provides at least three filter modes: low-pass filter mode, infrasound attenuation mode, and high-pass filter mode. The equalization preset unit has multiple built-in equalization presets, each with eight adjustable points for adjusting specific frequencies, bandwidths, and gains. It also has save and adjust functions, allowing users to edit and save preset values ​​for later retrieval. The automatic equalization operation unit includes a near-field scanning function subunit, a far-field scanning function subunit, a calculation subunit, and an equalization generation subunit. The near-field scanning function subunit is used to scan and generate a near-field curve, the far-field scanning function subunit is used to scan and generate a far-field curve, the calculation subunit is used to calculate the required peak or valley filling data and generate an adjusted initial EQ curve, and the equalization generation subunit is used to generate the final EQ curve by scanning again based on the adjusted initial EQ curve.

[0012] Furthermore, a spatial sound field automatic correction system is provided to implement the spatial sound field automatic correction method described above. The system includes a signal acquisition unit, a signal pre-detection / amplification unit, a data signal processing unit, and a main control chip. The main control chip communicates with a test terminal via a Bluetooth module. The test terminal has an automatic equalization (EQ) application unit and is connected to the EQ application unit via the Bluetooth module. The EQ application unit transmits adjustment instructions to the main control chip via the Bluetooth module. Based on the instructions, the main control chip transmits automatic correction instructions to the data signal processing unit. The signal acquisition unit acquires audio signals, which are then detected, compared, and amplified by the signal pre-detection / amplification unit. The signal pre-detection / amplification unit is connected to the main control chip to send the processed signal to the main control chip. Based on this signal, the main control chip transmits automatic correction instructions to the data signal processing unit for automatic equalization operation.

[0013] In another aspect, the present invention provides an audio system, including an audio body containing a power amplifier unit and a speaker. The power amplifier unit amplifies and processes the received audio signal before transmitting it to the speaker for playback. The audio body also has the aforementioned spatial sound field automatic correction system. The power amplifier unit is connected to the data signal processing unit of the spatial sound field automatic correction system via a post-stage detection and processing module. The post-stage detection and processing module receives the audio signal after automatic equalization processing and performs conversion, buffering, filtering, and amplification processing. The processed audio signal is then sent to the power amplifier unit and played through the speaker.

[0014] In the aforementioned automatic spatial sound field correction method and system, near-field and far-field curves are obtained by scanning with a test terminal at the near-field position and a predetermined distance from the speaker, respectively. Then, a predetermined algorithm is used to suppress peaks or fill valleys in the far-field curve to obtain an adjusted initial EQ curve. A second scan of the far-field curve is then performed to obtain the final frequency response curve for correction. The entire automatic equalization adjustment process is achieved through Bluetooth communication between a mobile phone or other terminal and the speaker's main control chip. The equalization can be adjusted according to the user's personalized selection in a mobile APP. During operation, corrections can be made at any time according to needs and actual effects, thus truly achieving real-time dynamic adjustment and meeting the user's personalized needs. It can be widely used in various audio equipment and has broad application prospects. Attached Figure Description

[0015] Figure 1 This is a schematic flowchart of the automatic spatial sound field correction method according to an embodiment of the present invention.

[0016] Figure 2 This is a schematic diagram of the near-field curve scanned in the spatial sound field automatic correction method of this invention.

[0017] Figure 3 In the spatial sound field automatic correction method of this invention embodiment, Figure 2 A schematic diagram of the far-field curve of the new scan is added based on the previous one.

[0018] Figure 4 In the spatial sound field automatic correction method of this invention embodiment, Figure 3 Based on this, an adjustment curve diagram generated by the algorithm is added.

[0019] Figure 5 In the spatial sound field automatic correction method of this invention embodiment, Figure 4 Based on this, a schematic diagram of the far-field curve after a second scan is added.

[0020] Figure 6 This is a schematic diagram of the initial parameter setting interface of the automatic equalization (EQ) application unit in the spatial sound field automatic correction method of this invention.

[0021] Figure 7 This is a schematic diagram of the filter type selection unit interface of the automatic equalization (EQ) application unit in the spatial sound field automatic correction method of this invention.

[0022] Figure 8 This is a schematic diagram of the automatic equalization preset unit interface of the automatic equalization EQ application unit in the automatic spatial sound field correction method of this invention.

[0023] Figure 9 This is a schematic diagram of the parameter values ​​of eight preset adjustable points in the automatic equalization (EQ) application unit of the spatial sound field automatic correction method in this embodiment of the invention.

[0024] Figure 10 This is a schematic diagram of the framework structure of the automatic spatial sound field correction system according to an embodiment of the present invention.

[0025] Figure 11 This is a schematic diagram of the chip and circuit structure of the signal acquisition unit ADC in the spatial sound field automatic correction system according to an embodiment of the present invention.

[0026] Figure 12 This is a schematic diagram of the data signal processing unit DSP chip and its circuit structure in the spatial sound field automatic correction system according to an embodiment of the present invention.

[0027] Figure 13 This is a schematic diagram of the audio processing chip MCU circuit structure in the spatial sound field automatic correction system according to an embodiment of the present invention.

[0028] Figure 14 This is a schematic diagram of the circuit structure of the signal pre-detection / amplification unit OP1 in the automatic spatial sound field correction system of this invention.

[0029] Figure 15 This is a schematic diagram of the circuit structure of the Bluetooth module in the automatic spatial sound field correction system according to an embodiment of the present invention. Detailed Implementation

[0030] The technical solutions of the present invention will be clearly and completely described below with reference to the accompanying drawings of the embodiments of the present invention. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. All other embodiments obtained by those skilled in the art based on the embodiments of the present invention without creative effort are within the scope of protection of the present invention.

[0031] Please see Figure 1 This diagram illustrates a flow chart of an automatic spatial sound field correction method according to an embodiment of the present invention. This automatic spatial sound field correction method is used for the automatic correction of the EQ curve of an audio system within a space, and includes the following steps: S10 connects to the test terminal's automatic equalization EQ application unit and speakers via Bluetooth; S20: Near-field curves are obtained by scanning at the near-field position of the audio equipment using a test terminal. S30, at a predetermined distance from the speaker, the far-field curve is obtained through the first scan by the test terminal. The far-field curve and the near-field curve are two synchronous curves formed based on the same frequency range as the horizontal axis during the scan, so that they can be superimposed in the same coordinate system for synchronous adjustment and calculation. S40, using a predetermined algorithm, the far-field curve is subjected to corresponding peak suppression or valley filling to obtain the adjusted EQ curve; S50, the audio signal processing unit in the audio system processes the audio data based on the obtained initial EQ curve, and then obtains the far-field quadratic curve by a second scan through the test terminal at the same far-field position, thus obtaining the final frequency response curve for correction. Each of the above scanning steps is performed in the automatic equalization (EQ) application unit of the test terminal.

[0032] In step S10, the test terminal is preferably a portable electronic mobile terminal such as a mobile phone or tablet, to facilitate user connection to the speaker and automatic equalization control. The speaker has a built-in Bluetooth module to communicate with the mobile terminal and transmit user control commands or parameter setting commands.

[0033] Furthermore, the far-field curve and the near-field curve are generated in the same EQ curve graph. The far-field curve and the near-field curve are frequency response data curves formed based on the same test signal at positions near and far from the speaker. The near-field position is preferably within a range of ≤1m relative to the speaker. The test terminal is preferably a mobile terminal such as a mobile phone or tablet, specifically a microphone within the test terminal. Specifically, as shown in Figure 2 and... Figure 3 The images show the scanned near-field and far-field curves, respectively. This can be done within an automatic EQ application (such as the Auto EQ app). For example, clicking the "Near" button in the Auto EQ application will scan the near-field curve. Generally, the near-field curve is approximately equivalent to the ideal EQ curve, located about 10 centimeters from the speaker. Figure 2 As shown, in the Auto EQ interface of the automatic equalizer application unit, the curve (light blue) is the near-field curve. Then, the far-field curve is obtained by scanning. Specifically, in the automatic equalizer application unit, click the far-field scan button, i.e., "Target," to scan the far-field curve. The far field is the normal listening position, which can be a distance of several meters from the speakers in a room, for example, within a range of 2 meters or more. Figure 3 As shown, the curve located at the lower position in the Auto EQ interface of the automatic equalization EQ application unit is the far field curve (light green). That is, the far field curve is shifted down by a predetermined unit relative to the near field curve, and there are obvious peaks and troughs.

[0034] After obtaining the far-field and near-field curves, adjustments are made using a predetermined algorithm, which is implemented within the app. Specifically, the predetermined algorithm includes setting peak attenuation filters at and near each peak of the far-field curve, resulting in an attenuation of -3.5 to -1 dB within the frequency range of those peaks and near the curve; and setting Shelf filters at and near each trough of the far-field curve, resulting in a gain of 3 to 6 dB within the frequency range of those troughs and near the curve, with the gain decreasing linearly from the trough towards both sides. Figure 4 As shown, the curve obtained after adjustment using the predetermined algorithm in this embodiment is the next curve, which is... Figure 3 The frequency response curve is closely related to the far-field curve, which is a new frequency response curve generated after correction based on the far-field curve (as shown by the orange curve in the figure). All of the above curves are displayed in real time in the Auto EQ interface of the automatic equalization EQ application unit.

[0035] Furthermore, the predetermined algorithm includes setting a peak attenuation filter in the 50Hz-65Hz frequency band, so that the attenuation amplitude of the far-field curve in this frequency range is -3.5 to -1 dB, and the peak attenuation amplitude decreases linearly from 50Hz to 65Hz; setting a peak attenuation filter in the 30Hz-33Hz frequency band, so that the attenuation amplitude of the far-field curve in this frequency range is -3 to -1 dB; setting a low-frequency Shelf filter in the 33Hz-50Hz frequency band, so that the gain amplitude of the far-field curve in this frequency range is 3 to 6 dB, and the gain decreases linearly from the trough to the two adjacent frequency bands; and setting a high-frequency Shelf filter in the 100Hz-120Hz frequency band, so that the gain amplitude of the far-field curve in this frequency range is 1 to 3 dB, and the gain decreases linearly from the trough to the two adjacent frequency bands.

[0036] Furthermore, when obtaining the near-field curve, the maximum sound pressure level at each frequency point is measured using a test terminal at the near-field position of the speaker, and the data is stored in the Near[] array; when obtaining the far-field curve, the maximum sound pressure level at each frequency point is measured using a test terminal at a listening position at a predetermined distance from the speaker, and the data is stored in the Target[] array; the frequency range of the test is 20-120Hz, and the total length of each of the following arrays is 101-120Hz; the predetermined algorithm includes the following steps: (1) Standardization of benchmark data a. Calculate the average value Near_mid of the Near[] array using the following formula: Near_mid = Sum of maximum sound pressure levels in the near field / Total frequency range; b. Perform the following transformation on each value N_i in the Near[] array: N'_i = N_i - Near_mid, where i is the index of the corresponding frequency and the range of i is 0-100; (2) Standardization of target data a. Calculate the average value Target_mid of the Target[] array using the following formula: Target_mid = Sum of maximum sound pressure levels in the far field / Total frequency range; b. Perform the transformation on each value T_j in the Target[] array: T'_j = T_j - Target_mid, where j is the index of the corresponding frequency and the range of j is 0-100; (3) Shape difference analysis Based on the Near[] array and the Target[] array, the standardized spectral difference is calculated as follows: Diff_k = T'_k - N'_k, resulting in the Diff[] array, where k is the index of the corresponding frequency, ranging from 0 to 100, T'_k is the converted value T'_j of the Target[] array at the corresponding frequency, and N'_k is the converted value N'_i of the Near[] array at the corresponding frequency. (4) Key feature point extraction Scan the Diff[] array to extract feature points, and store the results in the Peak[] array. The feature points include peaks and troughs. (5) EQ parameter generation a. Select the M feature points with the largest amplitude in the Peak[] array, where M ranges from 6 to 12; b. Calculation rules: Obtain the starting index of Peak, denoted as Left, Peak, Right; Calculate the bandwidth (BW) and quality factor (Q) of the Peak. The bandwidth (BW) of the Peak is defined as follows: for a center frequency Fc, there exists a left-hand extreme frequency and a right-hand extreme frequency. The spectral difference (Diff_k) between the left-hand extreme frequency and the right-hand extreme frequency is close to 0. The bandwidth (BW) is equal to the absolute value of the difference between the left-hand and right-hand extreme frequencies. The formula for calculating the quality factor (Q) is as follows: Q = Fc / BW; G = (Diff[Left]+Diff[Right]) / 2 - Diff[Peak], where Diff[Left] is the spectral difference of the frequency points to the left of the peak or valley, Diff[Right] is the spectral difference of the frequency points to the right of the peak or valley, and Diff[Peak] is the spectral difference of the current peak or valley point; (6) Output: Output M sets of EQ parameters, each set of EQ parameters includes center frequency Fc, gain value G, and Q value.

[0037] When calculating the quality factor Q, since the feature points of the Peak[] array include peaks and troughs, the spectral difference within a predetermined frequency range on both sides of the peak or trough is smaller the further away from the peak or trough. Therefore, there must exist frequency points spanning a predetermined length of the center frequency, where the Diff_k is close to 0, which is the extreme frequency point. In this embodiment of the invention, the M feature points with the largest amplitude of the Peak[] array are selected, and the frequency range of the predetermined band BW is extended with the peak or trough frequency as the center frequency for adjustment. Taking M=8 as an example, the maximum number of feature points processed by the data signal processing unit DSP is set to 8. Each feature point has a center frequency Fc, which spreads to the left and right sides from the center frequency Fc until the Diff_k of the extreme frequencies on the left and right sides is close to 0. That is to say, within the frequency range of the predetermined band BW of these 8 center frequencies Fc, the spectral difference is large and needs to be corrected, while frequency points not belonging to these ranges do not need to be corrected, thereby saving processing time and cost, improving data processing efficiency, and ensuring higher fidelity.

[0038] Furthermore, obtaining the final frequency response curve for correction by testing the far-field quadratic curve a second time using the test terminal specifically includes the following steps: The obtained initial EQ curve is input into the data signal processing unit; wherein, the initial EQ curve or EQ parameter data is first obtained by communicating with the test terminal through a main control chip, and then transmitted to the transceiver data signal processing unit by the main control chip; Based on the obtained initial EQ curve, the data signal processing unit processes the audio data of the frequency band within the band BW of each center frequency according to a corresponding set of EQ parameters and plays it. Then, at the same far-field location, the frequency response curve is obtained by measuring with a test terminal, resulting in the final frequency response curve used for calibration.

[0039] Furthermore, the step of applying a predetermined algorithm to suppress peaks or fill valleys in the far-field curve includes constructing an attenuation gain curve according to the predetermined algorithm, and then superimposing the attenuation gain curve with the far-field curve to obtain the adjusted initial EQ curve.

[0040] Furthermore, the automatic EQ application unit includes a phase adjustment unit, a delay adjustment unit, a polarity adjustment unit, a filter type selection unit, an EQ preset unit, an automatic EQ operation unit, and an automatic EQ interface unit. Each unit allows users to perform personalized EQ settings on the terminal. The phase adjustment unit is used for phase adjustment, with a phase adjustment range of 0º-180º; the delay adjustment unit is used for delay adjustment, with a delay adjustment range of 0-20ms; the polarity adjustment unit is used for positive and negative polarity adjustment, switching between positive and negative polarities, changing from 0º to 180º when activated; the filter type selection unit provides at least three filter modes: low-pass filter mode, infrasound attenuation mode, and high-pass filter mode. Filter mode; the equalization preset unit has multiple built-in equalization presets, each with 8 adjustable points for adjusting specific frequencies, bandwidths, and gains, and has saving and adjustment functions. Users can edit and modify preset values ​​and save them for later use; the automatic equalization operation unit has a near-field scanning function subunit, a far-field scanning function subunit, a calculation subunit, and an equalization generation subunit. The near-field scanning function subunit is used to scan and generate near-field curves, the far-field scanning function subunit is used to scan and generate far-field curves, the calculation subunit is used to calculate the required peak suppression or valley filling data and generate the adjusted initial EQ curve; the equalization generation subunit is used to generate the final EQ curve by scanning again based on the adjusted initial EQ curve. Figures 2 to 5 As shown, the example is the interface of the automatic EQ operation unit. You can enter this automatic EQ operation unit by clicking "AutoEQ". The near field scanning function subunit is operated through the "Near" button, the far field scanning function subunit is operated through the "Target" button, the calculation subunit is operated through the "Compute" button, and finally, the EQ generation subunit is operated through the "EQTarget" button. The buttons of these function units are on the same interface, which makes it easy to switch between operations. In addition, the curve examples are in the same coordinate system to more intuitively see the changes before and after adjustment. Figures 2-9 These are the contents displayed on each sub-interface of the automatic balancing interface unit.

[0041] Specifically, such as Figure 6 As shown, the automatic equalizer (EQ) application unit can be preset after logging in, i.e., configured in the Main interface. "Volume" represents volume adjustment, allowing you to control the machine's volume; "Phase" represents phase adjustment, adjustable between 0º-180º; "Delay" represents delay adjustment, adjustable between 0-20ms; "Polarity" represents polarity adjustment, changing the polarity from 0º to 180º when enabled; and "Night Mode" represents night mode adjustment, limiting the power to approximately 60W when night mode is enabled.

[0042] Furthermore, during automatic calibration, the cutoff frequency and slope are adjusted by selecting a predetermined filter. The predetermined filter includes a low-pass filter mode, an infrasound attenuation mode, and a high-pass filter mode. The slope adjustment range is preferably -12 to +12 dB / oct, more preferably -6 to +6 dB / oct, and the slope decreases as it approaches the cutoff frequency. The predetermined filter has multiple preset adjustment points, each with a corresponding preset adjustment value, including a frequency point, bandwidth, and gain. During automatic calibration, any preset adjustment point can be selectively selected. Each adjustment point has n frequency bands Freq, n bandwidths Q (i.e., quality factor Q), and n gain ranges. Each frequency band Freq corresponds to a bandwidth Q and a gain range. The maximum adjustable bandwidth Q range for each frequency band is 1-15, and the maximum adjustable gain range for each frequency band is -18 dB to +6 dB.

[0043] like Figure 7 As shown in the example, the filter type selection unit of the automatic equalization (EQ) application unit provides three filter modes: low-pass filter mode, subsonic attenuation mode, and high-pass filter mode. All three modes allow adjustment of the cutoff frequency and slope. The frequency adjustment range is 20Hz to 200Hz, and the slope adjustment is done in steps; the example in the figure shows four levels: 12, 24, 36, and 48.

[0044] like Figure 8 and Figure 9 As shown in the example, the automatic EQ application unit includes an EQ preset unit, which is the EQ menu in the interface. Clicking it enters the EQ preset settings interface. In this embodiment, the EQ preset unit has 6 built-in EQ presets, and each EQ preset example has 8 adjustable points. Figure 8 There are three preset parameters available for selection, which are used to adjust the specific frequency Freq, bandwidth Q, and gain Gain. The selectable value range for frequency Freq is 0-100Hz, the selectable value range for bandwidth Q is 1-15, and the selectable value range for gain Gain is -18dB to +6dB. Figure 9 Taking Preset 1, one of the six presets, as an example, each equalizer preset has eight adjustable points. The equalizer preset unit also has a save and adjustment function. Users can edit and modify the preset values ​​and save them for later use. After saving the preset modifications or directly calling the default preset, simply click the Load button to load the preset values ​​and apply them to the current equalizer calibration. The subsequent automatic EQ calibration process will be based on the loaded preset values.

[0045] Please see Figure 10 This invention also provides an automatic spatial sound field correction system, including a signal acquisition unit (ADC), a signal pre-detection / amplification unit (OP1), a data signal processing unit (DSP), and a main control chip (MCU). The MCU communicates with a test terminal via a Bluetooth module, for example, a mobile terminal such as a mobile phone. The test terminal has an automatic equalization (EQ) application unit, which is connected via Bluetooth. The ADC acquires audio signals, which are then detected, compared, and amplified by the OP1. The OP1 is connected to the MCU to send the processed signal. The MCU then transmits the automatic correction command to the DSP for automatic equalization, thus achieving the automatic spatial sound field correction method described above and adjusting the EQ.

[0046] Specifically, the signal acquisition unit (ADC) converts the three analog audio signals from L (left), R (right), and LFE (low-frequency effect) into high-precision digital signals (I2S format) for processing by the data signal processing unit (DSP). Simultaneously, it outputs one analog signal to the signal pre-detection / amplification unit OP1 (LM358) for signal detection. The signal pre-detection / amplification unit OP1 compares / amplifies the analog signal output from the ADC to perform signal detection, determining if there is valid audio input. The detection result is then transmitted to the main control chip (MCU) for system wake-up, standby control, or status indication.

[0047] The data signal processing unit (DSP) is the core unit of the AutoEQ in this system. It receives digital audio from the signal acquisition unit (ADC) via I2S and performs automatic equalization correction based on commands from the main control chip (MCU), namely calibration instructions, various user-set parameters or preset parameters, and then transmits the status back.

[0048] After automatic dynamic equalization (Auto EQ) correction, the processed digital audio signal is sent to the OP2 stage (shown as a TL072 device) for digital-to-analog conversion and buffering, then processed by the power amplifier, and finally played through the speakers.

[0049] The spatial sound field automatic correction system also includes a trigger switch unit (SWITCH) with a standby time set to 20 minutes. When switched to "Always on," the device (speaker) will not enter standby mode after 20 minutes. Switching to "Auto on" will automatically put the device into standby mode after 20 minutes of no signal input. Switching to "Trigger Only" will force the device into standby mode, requiring a 12V voltage supplied through the 3.5mm headphone jack to wake it up. It also features LED indicators, a potentiometer (VR) reader, and USB-A (debugging / communication) input connections. Human-machine interaction includes reading VR (potentiometer), SWITCH (button), and USB-A (debugging / communication) inputs. The system connects the app to the speaker via Bluetooth, configures the DSP signal processing unit via I2C, and can also perform firmware upgrades or parameter adjustments via USB. In addition, the system also includes a power management system, which has a DC-DC transfer module to convert the bus voltage into a stable low-voltage DC (such as +5V, +12V, -12V, etc.) to power the circuits of the signal acquisition unit ADC, the signal pre-detection / amplification unit OP1, the data signal processing unit DSP, the main control chip MCU, etc.

[0050] Preferably, such as Figure 11 As shown, the signal acquisition unit ADC uses a PCM1861DBT single chip, supporting four differential analog audio inputs to simultaneously acquire left and right channel line inputs; as shown, there are four sets of left and right channel line input pins VINL1-VINL4 and VINR1-VINR4. Furthermore, the signal acquisition unit ADC has a built-in microphone bias port (MicBias) and a reference voltage module to internally drive the microphone input. Specifically, the signal acquisition unit ADC has an analog source pin AVDD, a digital source pin DVDD, and an IO source pin IOVDD. The analog source pin AVDD is connected to the power supply through a tenth inductor L10, and a first set of multi-stage decoupling capacitors is connected between the analog source pin AVDD and the tenth inductor L10; the digital source pin DVDD and the IO source pin IOVDD are both connected to the power supply through a twelfth inductor L12, and a second set of multi-stage decoupling capacitors is connected between the twelfth inductor L12 and the chip pins.

[0051] Furthermore, such as Figure 12As shown, the data signal processing unit (DSP) adopts the ADAU1701 chip U10 with integrated SigmaDSP architecture, which has a built-in 28 / 56-bit audio processing core, 2 ADC input terminals and 4 DAC output terminals to realize a complete link of analog audio acquisition, digital signal processing and audio output. The data signal processing unit (DSP) adopts a separate power supply architecture for analog power supply AVDD and digital power supply DVDD, and has multiple decoupling capacitors and power enable MOSFET Q8 to effectively suppress crosstalk of digital switching noise to the analog audio path.

[0052] Preferably, such as Figure 13 As shown, the audio processing chip MCU adopts the NUC029LEE microcontroller U19, which integrates an I2C interface, a UART interface, an SPI interface, a USB interface, and multiple ADC / GPIO interfaces. It sends EQ parameters and configuration commands to the data signal processing unit DSP through the I2C interface, realizes Bluetooth communication through the UART interface, and realizes power supply detection and data interaction through the USB interface.

[0053] Preferably, such as Figure 14As shown, the signal pre-detection / amplification unit OP1 adopts a dual operational amplifier circuit structure, including a differential amplifier unit, a second-order low-pass filter unit, and a precision rectification unit. The differential amplifier unit includes a left channel input line, a right channel input line, a differential amplifier U1-A, and a feedback line. The left channel input line and the right channel input line enter the inverting input of the differential amplifier U1-A through corresponding input resistors and filter capacitors. As shown in the figure, the left channel input line has a twenty-first input resistor R21 and a ninetieth filter capacitor C90, and the right channel input line has a twenty-eighth input resistor R218 and a twenty-fourth filter capacitor C224. The feedback circuit includes a parallel feedback resistor R215 and a feedback capacitor C223 connected between the inverting input and the output of the differential amplifier U1-A. The second-order low-pass filter unit includes a voltage-following low-pass filter U1-B. The output of the differential amplifier U1-A is connected to the inverting input of the low-pass filter U1-B via a 9th capacitor C91 and a 213 resistor R213. The non-inverting inputs of the differential amplifier U1-A and the low-pass filter U1-B are connected to the power supply via corresponding current-limiting resistors. For example, the non-inverting input of the differential amplifier U1-A is connected to the power supply via a 216 resistor current-limiting resistor R216, and the non-inverting input of the low-pass filter U1-B is connected to the power supply via a 214 resistor current-limiting resistor R214. The power supply is +3.3V. The front-end line connecting the power supply to the two in-phase terminals is equipped with multi-stage decoupling capacitors (such as C226, C227, C228, C230) and a precision rectified peak detection amplifier U1-C. The peak detection amplifier U1-C is used to convert AC audio signals into DC voltage. The output terminal of the low-pass filter U1-B is connected to the acquisition pin of the audio processing chip MCU, which is also connected to the acquisition pin of the audio processing chip MCU via capacitor C93 and resistor R24.

[0054] Preferably, such as Figure 15 As shown, the Bluetooth module includes a Bluetooth chip BT1, an RF antenna ANT1, and a Bluetooth power management unit. The Bluetooth chip BT1 has a UART interface, a USB interface, and an analog DAC interface. The Bluetooth chip BT1 communicates with the audio processing chip MCU via the UART interface, performs audio transmission and firmware upgrades via the USB interface, and outputs left and right channel audio signals via the analog DAC interface. The Bluetooth power management unit is connected to the power pin VBAT of the Bluetooth chip BT1. The Bluetooth power management unit includes an input terminal VCC connected to the input source, a ninth inductor L9 connected to the input terminal VCC, and multiple sets of filter capacitors (such as C30, C2, and C3) connected between the input terminal VCC and the ninth inductor L9. The Bluetooth chip BT1 is preferably an MD-BT-AC6951B chip.

[0055] In another aspect, the present invention provides an audio system, including an audio body containing a power amplifier unit and a speaker. The power amplifier unit amplifies and processes received audio signals before transmitting them to the speaker for playback. The audio body also has the aforementioned automatic spatial sound field correction system. Figure 10 As shown, the power amplifier unit AMP is connected to the data signal processing unit DSP in the spatial sound field automatic correction system via a post-stage detection and processing module OP2 (TL072). The post-stage detection and processing module OP2 (TL072) is used to receive the audio signal after automatic equalization processing, and perform conversion, buffering, filtering and amplification processing. The processed audio signal is sent to the power amplifier unit and then played through the speaker.

[0056] It should be noted that the present invention is not limited to the above-described embodiments. Based on the inventive spirit of the present invention, those skilled in the art can make other changes, and these changes made in accordance with the inventive spirit of the present invention should be included within the scope of protection claimed by the present invention.

Claims

1. A method for automatic spatial sound field correction, used for automatic correction of the EQ curve of sound within a space, characterized in that, Includes the following steps: The test terminal's automatic equalization EQ application unit and speakers are connected via Bluetooth. Near-field curves were obtained by scanning at the near-field position of the audio equipment using a test terminal. At a predetermined distance from the speaker, the far-field curve is obtained through the first scan using the test terminal. The far-field curve and the near-field curve are two synchronous curves formed based on the same frequency range as the horizontal axis during the scan, so that they can be superimposed in the same coordinate system for synchronous adjustment and calculation. By using a predetermined algorithm, the far-field curve is subjected to corresponding peak suppression or valley filling to obtain the adjusted EQ curve; The data signal processing unit in the audio system processes the audio data based on the obtained initial EQ curve, and then obtains the far-field quadratic curve by a second scan through the test terminal at the same far-field position, thus obtaining the final frequency response curve for correction. Each of the above scanning steps is performed in the automatic equalization (EQ) application unit of the test terminal.

2. The automatic spatial sound field correction method as described in claim 1, characterized in that, The far-field curve and the near-field curve are generated in the same EQ curve graph. The far-field curve and the near-field curve are frequency response data curves formed at positions close to the speaker and far from the speaker, respectively, based on the same test signal. The near-field curve is generated by measuring the maximum sound pressure level at each frequency point using a microphone at the near-field position of the speaker. The far-field curve is generated by measuring the maximum sound pressure level at each frequency point using a listening position at a predetermined distance from the speaker.

3. The automatic spatial sound field correction method as described in claim 1, characterized in that, The predetermined algorithm includes setting peak attenuation filters at and near each peak of the far-field curve, so that the attenuation of the far-field curve at and near the peak is -3.5 to -1 dB in the frequency range at and near the peak, and setting Shelf filters at and near each trough of the far-field curve, so that the gain of the far-field curve at and near the trough is 3 to 6 dB in the frequency range at and near the trough, and the gain decreases linearly from the trough to both sides of the frequency band.

4. The automatic spatial sound field correction method as described in claim 1, characterized in that, The predetermined algorithm includes setting a peak attenuation filter in the 50Hz-65Hz frequency band, so that the attenuation amplitude of the far-field curve in this frequency range is -3.5 to -1 dB, and the peak attenuation amplitude decreases linearly from 50Hz to 65Hz; setting a peak attenuation filter in the 30Hz-33Hz frequency band, so that the attenuation amplitude of the far-field curve in this frequency range is -3 to -1 dB; setting a low-frequency Shelf filter in the 33Hz-50Hz frequency band, so that the gain amplitude of the far-field curve in this frequency range is 3 to 6 dB, and the gain decreases linearly from the trough to the two adjacent frequency bands; and setting a high-frequency Shelf filter in the 100Hz-120Hz frequency band, so that the gain amplitude of the far-field curve in this frequency range is 1 to 3 dB, and the gain decreases linearly from the trough to the two adjacent frequency bands.

5. The automatic spatial sound field correction method as described in claim 1, characterized in that, When obtaining the near-field curve, the maximum sound pressure level at each frequency point is measured using a test terminal at the near-field position of the speaker, and the data is stored in the Near[] array; when obtaining the far-field curve, the maximum sound pressure level at each frequency point is measured using a test terminal at a listening position at a predetermined distance from the speaker, and the data is stored in the Target[] array; the frequency range of the test is 20-120Hz, and the total length of each of the following arrays is 101-120Hz; the predetermined algorithm includes the following steps: (1) Standardization of benchmark data a. Calculate the average value Near_mid of the Near[] array using the following formula: Near_mid = Sum of maximum sound pressure levels in the near field / Total frequency range; b. Perform the following transformation on each value N_i in the Near[] array: N'_i = N_i - Near_mid, where i is the index of the corresponding frequency and the range of i is 0-100; (2) Standardization of target data a. Calculate the average value Target_mid of the Target[] array using the following formula: Target_mid = Sum of maximum sound pressure levels in the far field / Total frequency range; b. Perform the transformation on each value T_j in the Target[] array: T'_j = T_j - Target_mid, where j is the index of the corresponding frequency and the range of j is 0-100; (3) Shape difference analysis Based on the Near[] array and the Target[] array, the standardized spectral difference is calculated as follows: Diff_k = T'_k - N'_k, resulting in the Diff[] array, where k is the index of the corresponding frequency, ranging from 0 to 100, T'_k is the converted value T'_j of the Target[] array at the corresponding frequency, and N'_k is the converted value N'_i of the Near[] array at the corresponding frequency. (4) Key feature point extraction Scan the Diff[] array to extract feature points, and store the results in the Peak[] array. The feature points include peaks and troughs. (5) EQ parameter generation a. Select the M feature points with the largest amplitude in the Peak[] array, where M ranges from 6 to 12; b. Calculation rules: Obtain the starting index of Peak, denoted as Left, Peak, Right; Calculate the bandwidth (BW) and quality factor (Q) of the Peak. The bandwidth (BW) of the Peak is defined as follows: for a center frequency Fc, there exists a left-hand extreme frequency and a right-hand extreme frequency. The spectral difference (Diff_k) between the left-hand extreme frequency and the right-hand extreme frequency is close to 0. The bandwidth (BW) is equal to the absolute value of the difference between the left-hand and right-hand extreme frequencies. The formula for calculating the quality factor (Q) is as follows: Q = Fc / BW; G = (Diff[Left]+Diff[Right]) / 2 - Diff[Peak], where Diff[Left] is the spectral difference of the frequency points to the left of the peak or valley, Diff[Right] is the spectral difference of the frequency points to the right of the peak or valley, and Diff[Peak] is the spectral difference of the current peak or valley point; (6) Output: Output M sets of EQ parameters, each set of EQ parameters includes center frequency Fc, gain value G, and Q value.

6. The automatic spatial sound field correction method as described in claim 5, characterized in that, The process of obtaining the final frequency response curve for correction by testing the far-field quadratic curve a second time using the terminal includes the following steps: The obtained initial EQ curve is input into the data signal processing unit; wherein, the initial EQ curve or EQ parameter data is obtained by communicating with the test terminal through a main control chip, and then transmitted to the transceiver data signal processing unit by the main control chip; Based on the obtained initial EQ curve, the data signal processing unit processes the audio data of the frequency band within the band BW of each center frequency according to a corresponding set of EQ parameters and plays it. Then, at the same far-field location, the frequency response curve is obtained by measuring with a test terminal, resulting in the final frequency response curve used for calibration.

7. The automatic spatial sound field correction method as described in claim 1, characterized in that, The step of applying a predetermined algorithm to suppress peaks or fill valleys in the far-field curve includes constructing an attenuation gain curve according to the predetermined algorithm, and then superimposing the attenuation gain curve with the far-field curve to obtain the adjusted initial EQ curve. The automatic equalization EQ application unit is equipped with a phase adjustment unit, a delay adjustment unit, a polarity adjustment unit, a filter type selection unit, an equalization preset unit, an automatic equalization operation unit, and an automatic equalization interface unit. Each unit is used by the user to perform personalized equalization settings on the terminal. The phase adjustment unit is used for phase adjustment, and the phase adjustment range is 0º-180º. The delay adjustment unit is used for delay adjustment, with a delay adjustment range of 0-20ms; the polarity adjustment unit is used for positive and negative polarity adjustment, switching between positive and negative polarities, changing from 0º to 180º when activated; the filter type selection unit provides at least three filter modes: low-pass filter mode, infrasound attenuation mode, and high-pass filter mode; the equalization preset unit has multiple built-in equalization presets, each with 8 adjustable points, used to adjust specific frequencies, bandwidth, and gain, and has saving and adjustment functions, allowing users to edit and modify preset values ​​and save them for later use; the automatic equalization operation unit has a near-field scanning function subunit, a far-field scanning function subunit, a calculation subunit, and an equalization generation subunit. The near-field scanning function subunit is used to scan and generate a near-field curve, the far-field scanning function subunit is used to scan and generate a far-field curve, the calculation subunit is used to calculate the required peak or valley filling data and generate an adjusted initial EQ curve; the equalization generation subunit is used to scan again based on the adjusted initial EQ curve to generate the final EQ curve.

8. The automatic spatial sound field correction method as described in claim 1, characterized in that, During automatic calibration, the cutoff frequency and slope are adjusted by selecting a predetermined filter. The predetermined filter includes a low-pass filter mode, an infrasound attenuation mode, and a high-pass filter mode. The slope adjustment range is -6 to +6 dB / oct, and the slope decreases as it approaches the cutoff frequency. The predetermined filter has multiple preset adjustment points, each with a corresponding preset adjustment value, including a frequency point, bandwidth, and gain. During automatic calibration, any preset adjustment point can be selectively selected. Each adjustment point has n frequency bands (Freq), n bandwidths (Q), and n gain ranges. Each frequency band (Freq) corresponds to a bandwidth (Q) and a gain range. The maximum adjustable bandwidth (Q) range for each frequency band is 1-15, and the maximum adjustable gain range for each frequency band is -18 dB to +6 dB.

9. A spatial sound field automatic correction system for implementing the spatial sound field automatic correction method as described in any one of claims 1-8, the system comprising a signal acquisition unit, a signal pre-detection / amplification unit, a data signal processing unit, and a main control chip, characterized in that, The main control chip communicates with a test terminal via a Bluetooth module. The test terminal has an automatic equalization (EQ) application unit and communicates with the EQ application unit via the Bluetooth module. The EQ application unit transmits adjustment instructions to the main control chip via the Bluetooth module. Based on the instructions, the main control chip transmits the automatic correction instructions to the data signal processing unit. The signal acquisition unit acquires audio signals, which are then detected, compared, and amplified by the signal pre-detection / amplification unit. The signal pre-detection / amplification unit is connected to the main control chip to send the processed signal to the main control chip. Based on this signal, the main control chip transmits the automatic correction instructions to the data signal processing unit for automatic equalization.

10. A speaker, comprising a speaker body, wherein the speaker body contains a power amplifier unit and a speaker, the power amplifier unit amplifies and processes received audio signals and transmits them to the speaker for playback; characterized in that, The speaker body also has a spatial sound field automatic correction system as described in claim 9. The power amplifier unit is connected to the data signal processing unit in the spatial sound field automatic correction system via a post-stage detection and processing module. The post-stage detection and processing module is used to receive the audio signal after automatic equalization processing, and perform conversion, buffering, filtering and amplification processing. The processed audio signal is sent to the power amplifier unit and then played through the speaker.

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