Audio calibration method applied to distributed television audio system and related device

Abstract

This application belongs to the field of audio processing technology and relates to an audio calibration method and related equipment applied to a distributed television audio system. The method includes: performing spatial topology identification on a television host and several audio devices placed in a target space to obtain three-dimensional spatial data of the audio devices relative to the television host; calculating the orientation information of the audio devices based on the three-dimensional spatial data; performing phase calibration processing on the phase parameters of the audio stream of the audio devices based on the orientation information; performing delay calibration processing on the delay parameters of the audio stream based on the orientation information; and performing channel calibration processing on the channel allocation of the audio stream based on the orientation information. This application can accurately identify the position information of the audio devices relative to the television host and accurately calibrate the audio stream of the audio devices based on this information, thereby improving the audio playback quality of the distributed television audio system and providing users with a higher quality and more realistic listening experience.

Description

Technical Field

[0001] This application relates to the field of audio processing technology, and in particular to audio calibration methods and related equipment applied to distributed television audio systems. Background Technology

[0002] With the continuous development of television technology, distributed television audio systems have gradually become mainstream. A distributed television audio system typically consists of a television host placed in the target space and several audio devices distributed in different locations, working together to achieve rich audio effects such as surround sound.

[0003] However, in practical applications, due to the different relative positions of the audio devices and the TV host, and the influence of various factors such as distance and obstacles on the audio signal during transmission, the audio playback quality is severely affected, making it impossible for users to obtain an ideal listening experience.

[0004] This shows that traditional distributed television audio systems have the problem of audio playback quality failing to meet user needs. Summary of the Invention

[0005] The purpose of this application is to propose an audio calibration method and related equipment for distributed television audio systems, so as to solve the problem that the audio playback quality of traditional distributed television audio systems cannot meet the needs of users.

[0006] To address the aforementioned technical problems, this application provides an audio calibration method for a distributed television audio system, employing the following technical solution: A spatial topology recognition operation is performed on the TV host and several audio devices placed in the target space to obtain the three-dimensional spatial data of the audio devices relative to the TV host. Calculate the orientation information of the audio device based on the three-dimensional spatial data; The audio stream of the audio device is calibrated based on the location information. The step of performing audio calibration processing on the audio stream of the audio device based on the orientation information specifically includes the following steps: The phase parameters of the audio stream are calibrated based on the azimuth information. The delay parameters of the audio stream are calibrated based on the location information. Based on the location information, the channel allocation of the audio stream is calibrated.

[0007] To address the aforementioned technical problems, this application also provides an audio calibration device for a distributed television audio system, employing the following technical solution: The spatial topology recognition module is used to perform spatial topology recognition on a TV host and several audio devices placed in the target space, and obtain three-dimensional spatial data of the audio devices relative to the TV host. A location information calculation module is used to calculate the location information of the audio device based on the three-dimensional spatial data. An audio calibration module is used to perform audio calibration processing on the audio stream of the audio device based on the orientation information; The audio calibration module includes: a phase calibration submodule, a delay calibration submodule, and a channel calibration submodule, wherein: The phase calibration submodule is used to perform phase calibration processing on the phase parameters of the audio stream based on the orientation information; The delay calibration submodule is used to perform delay calibration processing on the delay parameters of the audio stream based on the orientation information; The channel calibration submodule is used to perform channel calibration processing on the channel allocation of the audio stream based on the orientation information.

[0008] To address the aforementioned technical problems, this application also provides a computer device that employs the following technical solution: It includes a memory and a processor, wherein the memory stores computer-readable instructions, and the processor executes the computer-readable instructions to implement the steps of the audio calibration method applied to a distributed television audio system as described above.

[0009] To address the aforementioned technical problems, this application also provides a computer-readable storage medium, employing the technical solution described below: The computer-readable storage medium stores computer-readable instructions, which, when executed by a processor, implement the steps of the audio calibration method applied to a distributed television audio system as described above.

[0010] This application provides an audio calibration method for a distributed television audio system, comprising: performing spatial topology identification on a television host and several audio devices placed in a target space to obtain three-dimensional spatial data of the audio devices relative to the television host; calculating the orientation information of the audio devices based on the three-dimensional spatial data; and performing audio calibration processing on the audio stream of the audio devices based on the orientation information. The step of performing audio calibration processing on the audio stream of the audio devices based on the orientation information specifically includes the following steps: performing phase calibration processing on the phase parameters of the audio stream based on the orientation information; performing delay calibration processing on the delay parameters of the audio stream based on the orientation information; and performing channel calibration processing on the channel allocation of the audio stream based on the orientation information. Compared with existing technologies, this application can accurately identify the position information of the audio devices relative to the television host and perform precise calibration on the audio stream of the audio devices based on this information, thereby improving the audio playback quality of the distributed television audio system and providing users with a higher quality and more realistic listening experience. Attached Figure Description

[0011] To more clearly illustrate the solutions in this application, the accompanying drawings used in the description of the embodiments of this application will be briefly introduced below. Obviously, the accompanying drawings described below are some embodiments of this application. For those skilled in the art, other drawings can be obtained from these drawings without creative effort.

[0012] Figure 1 This is an exemplary system architecture diagram to which this application can be applied; Figure 2 This is a flowchart illustrating the implementation of an audio calibration method for a distributed television audio system provided in this application embodiment; Figure 3 This is a schematic diagram of the structure of an audio calibration device for a distributed television audio system provided in an embodiment of this application; Figure 4 This is a schematic diagram of the structure of one embodiment of the computer device according to this application. Detailed Implementation

[0013] Unless otherwise defined, all technical and scientific terms used herein have the same meaning as commonly understood by one of ordinary skill in the art to which this application pertains; the terminology used herein in the specification of the application is for the purpose of describing particular embodiments only and is not intended to be limiting of the application; the terms "comprising" and "having," and any variations thereof, in the specification, claims, and foregoing drawings of this application, are intended to cover non-exclusive inclusion. The terms "first," "second," etc., in the specification, claims, or foregoing drawings of this application are used to distinguish different objects, not to describe a particular order.

[0014] In this document, the term "embodiment" means that a particular feature, structure, or characteristic described in connection with an embodiment may be included in at least one embodiment of this application. The appearance of this phrase in various places throughout the specification does not necessarily refer to the same embodiment, nor is it a separate or alternative embodiment mutually exclusive with other embodiments. It will be explicitly and implicitly understood by those skilled in the art that the embodiments described herein can be combined with other embodiments.

[0015] To enable those skilled in the art to better understand the present application, the technical solutions in the embodiments of the present application will be clearly and completely described below with reference to the accompanying drawings.

[0016] like Figure 1 As shown, the distributed television audio system 100 may include a television host 101 and several audio devices 102. The television host 101 is the core control and processing device of the distributed television audio system. This host can receive audio and video signals from various signal sources, perform preliminary processing on the audio signals (such as decoding different audio encoding formats (e.g., Dolby Digital, DTS), converting them into digital audio signals that the system can further process and transmit). Then, the processed audio signals are distributed to the various audio devices. In addition, the television host is equipped with a user interface, allowing users to interact with the host via remote control, mobile app, etc., to perform various settings and controls on the distributed television audio system. For example, users can adjust the volume, select different audio modes (e.g., movie mode, music mode, game mode), and perform audio calibration operations. The audio devices 102 refer to various electronic devices or apparatuses used for processing, generating, playing, recording, or transmitting sound signals, with their core functions revolving around sound acquisition, processing, output, and storage.

[0017] It should be noted that the audio calibration method for a distributed television audio system provided in this application embodiment is generally executed by the television host 101, and correspondingly, the audio calibration device for a distributed television audio system is generally located in the television host 101.

[0018] It should be understood that Figure 1 The number of television hosts 101 and audio devices 102 shown is merely illustrative. Any number of television hosts 101 and audio devices 102 can be used depending on implementation requirements.

[0019] Continue to refer to Figure 2The diagram shows a flowchart of an embodiment of an audio calibration method for a distributed television audio system according to this application. The aforementioned audio calibration method for a distributed television audio system includes steps S201, S202, and S203.

[0020] In step S201, a spatial topology recognition operation is performed on the TV host and several audio devices placed in the target space to obtain three-dimensional spatial data of the audio devices relative to the TV host.

[0021] In this embodiment, the target space refers to a specific physical space where a television host and several audio devices are placed, and where audio calibration is required to achieve good audio playback. This target space can be an indoor space of various sizes and shapes, ranging from a small family living room where people watch television programs and movies through a distributed television audio system and enjoy an immersive audio experience, to a large professional audio-visual room, theater, or conference room. For example, in a theater, by reasonably arranging audio equipment and using this audio calibration method, audiences in different seats can obtain high-quality audio effects. For example, in a conference room, it can ensure that participants can clearly hear the content of the speech.

[0022] In this application embodiment, specific sensors, such as ultrasonic sensors and infrared sensors, can be installed on the TV host and audio equipment to measure the distance and angle information between the devices by transmitting and receiving signals. Alternatively, this application can employ computer vision-based technology to capture images of the target space through a camera, and then use image processing algorithms to identify the positions of the TV host and audio equipment, thereby calculating the relative three-dimensional spatial data between them, including horizontal distance, vertical distance, and azimuth angle. It should be understood that the example of spatial topology recognition operation here is only for ease of understanding and is not intended to limit this application.

[0023] In step S202, the orientation information of the audio device is calculated based on the three-dimensional spatial data.

[0024] In some optional implementations of the embodiments of this application, the orientation information includes the actual straight-line distance between the audio device and the television host, the azimuth angle, and the pitch angle, wherein the actual straight-line distance... Represented as:

[0025] in, , , These represent the coordinate data of the three-dimensional spatial data; Azimuth Represented as:

[0026] Pitch angle Represented as: .

[0027] In step S203, the audio stream of the audio device is calibrated according to the orientation information.

[0028] In this embodiment of the application, step S203 specifically includes the following steps: step S2031, step S2032, and step S2033, wherein: In step S2031, phase calibration processing is performed on the phase parameters of the audio stream of the audio device based on the orientation information.

[0029] In this embodiment, due to the varying distances between the audio devices and the television host, the arrival times of the sound signals at each audio device differ, resulting in phase inconsistency. Based on the calculated azimuth information, the relative distance between each audio device and the television host can be determined, and the corresponding phase difference can be calculated. Then, by adjusting the phase of the audio signals in the audio devices, the audio signals output by each audio device are made to maintain phase consistency, avoiding sound distortion and interference caused by phase differences.

[0030] In step S2032, the delay parameters of the audio stream are calibrated based on the orientation information.

[0031] In the audio-visual system of this application embodiment, the synchronization of sound and image is crucial. However, due to the different paths and delays that audio and video signals may experience during transmission and processing, asynchrony between sound and image occurs, i.e., audio-visual desynchronization. This desynchronization affects the user's viewing experience, especially when watching movies, television programs, or making video calls. Therefore, this application calibrates the delay parameters of the audio stream based on the orientation information of the audio device and the television host. Orientation information typically refers to the positional relationship of the audio device relative to the television host, including distance, angle, etc. By obtaining this orientation information, the time required for the audio signal to travel from the audio device to the television host can be calculated, thereby adjusting the playback time of the audio stream to synchronize it with the video signal.

[0032] In step S2033, channel calibration processing is performed on the channel allocation of the audio stream based on the orientation information.

[0033] In this embodiment of the application, in a distributed television audio system, different audio devices are typically responsible for playing different channels, such as the left channel, right channel, center channel, and surround channels. Based on the location information, the optimal channel allocation for each audio device in the sound field can be accurately determined, ensuring that the sound of each channel can be played from the appropriate position, thereby achieving a clearer and more realistic surround sound effect.

[0034] This application provides an audio calibration method for a distributed television audio system, comprising: performing spatial topology identification on a television host and several audio devices placed in a target space to obtain three-dimensional spatial data of the audio devices relative to the television host; calculating the orientation information of the audio devices based on the three-dimensional spatial data; and performing audio calibration processing on the audio stream of the audio devices based on the orientation information. The step of performing audio calibration processing on the audio stream of the audio devices based on the orientation information specifically includes the following steps: performing phase calibration processing on the phase parameters of the audio stream based on the orientation information; performing delay calibration processing on the delay parameters of the audio stream based on the orientation information; and performing channel calibration processing on the channel allocation of the audio stream based on the orientation information. Compared with the prior art, this application can accurately identify the position information of the audio devices relative to the television host and perform precise calibration on the audio stream of the audio devices based on this information, thereby improving the audio playback quality of the distributed television audio system and providing users with a higher quality and more realistic listening experience.

[0035] In some optional implementations of the embodiments of this application, the aforementioned television host includes a signal transmitter, and the audio device includes a signal receiver. The step of performing spatial topology identification on the television host and several audio devices placed in the target space to obtain the three-dimensional spatial data of the audio devices relative to the television host specifically includes the following steps: The signal transmitter sends a low-frequency detection signal into the target space; The arrival time and strength of the signal are obtained by receiving the low-frequency detection signal from the signal receiver. The three-dimensional spatial data of the audio device relative to the TV host is calculated based on the signal arrival time and signal strength.

[0036] In this embodiment, the spatial topology recognition operation is mainly used to calculate the three-dimensional spatial data of the audio device relative to the television host. This spatial topology recognition operation can be: (1) The TV host sends low-frequency detection signals (infrasound or ultrasound) to surrounding slave devices (such as Bluetooth speakers, soundbars). Specifically, the signal transmitter built into the TV host sends low-frequency detection signals to the surrounding space at a fixed frequency to ensure that the signal can be received by all audio devices. (2) Each audio device receives the signal and feeds back the time difference of arrival (TDOA) and intensity difference of the signal to the host. Specifically, after the built-in signal receiver of the audio device receives the low-frequency detection signal sent by the host, it calculates the time difference of arrival and intensity difference of the signal through the built-in microprocessor, and feeds back the calculation result to the TV host through the wireless communication module. (3) The host uses an algorithm to calculate the three-dimensional spatial coordinates (distance, azimuth, and pitch) of each audio device relative to the TV. Specifically, the microprocessor built into the TV host calculates the three-dimensional spatial coordinates of each audio device relative to the TV based on the time difference and intensity difference fed back by the audio devices using a triangulation algorithm.

[0037] Compared with existing technologies, this application achieves three-dimensional spatial positioning with centimeter-level accuracy by analyzing signal arrival time (ToA) and intensity (RSSI).

[0038] In some optional implementations of the embodiments of this application, the step of calculating the three-dimensional spatial data of the audio device relative to the television host based on the signal arrival time and signal arrival strength specifically includes the following steps: The signal propagation time is calculated based on the transmission time and arrival time of the low-frequency detection signal; Calculate the predicted straight-line distance between the audio equipment and the television host based on the signal propagation time; Construct and predict the signal strength corresponding to the straight-line distance based on the model of signal strength attenuation with distance; The signal strength is converted into an auxiliary distance based on the fitted model; The predicted straight-line distance is weighted and fused based on the auxiliary distance to obtain the true straight-line distance; The three-dimensional spatial data of the audio device relative to the TV host are calculated based on the actual straight-line distance and the trilateration method.

[0039] In this embodiment, the signal propagation time is calculated based on the transmission time and arrival time of the low-frequency detection signal. ,in:

[0040] in, Indicates the first The signal arrival time of each audio device Indicates the first The transmission time of the low-frequency detection signal of an audio device.

[0041] In this embodiment of the application, based on the signal propagation time Calculate the predicted straight-line distance between the audio device and the TV host. ,in:

[0042] in, Indicates the speed of sound propagation ( 343m / s).

[0043] In this embodiment, a signal strength corresponding to the straight-line distance is constructed based on a model of signal strength attenuation with distance. ,in:

[0044] in, Indicates the reference distance. This represents the reference signal strength corresponding to the reference distance. This represents the path loss index. This represents shadow fading noise that follows a normal distribution.

[0045] In this embodiment of the application, the signal strength is determined according to the fitting model. Convert to auxiliary distance ,in:

[0046] in, Indicates the reference distance. This represents the reference signal strength corresponding to the reference distance. This represents the signal strength corresponding to the straight-line distance. This represents the path loss index.

[0047] In this embodiment of the application, based on the auxiliary distance For predicting straight-line distance We perform weighted fusion to obtain the true straight-line distance. ,in:

[0048] in, This indicates the weighted fusion weight.

[0049] In this embodiment, the three-dimensional spatial data of the audio device relative to the television host is calculated based on the actual straight-line distance and the trilateration method. Specifically: (1) Using the coordinates A( of the three transmitters on the TV host) , , ), B ( , , ), B ( , , ( ) represents the center of the sphere, and the true straight-line distance from the center of the sphere to the audio device. Construct three spherical equations for the radius:

[0050] (2) Solve the linear equations using geometric methods to obtain the three-dimensional spatial data (x, y, z) of the audio device relative to the TV host.

[0051] Compared with existing technologies, this application will assist in distance Distance results with signal arrival time (ToA) Weighted fusion is performed to effectively improve anti-interference capabilities and achieve highly robust 3D positioning.

[0052] In some optional implementations of this application, the step of performing phase calibration processing on the phase parameters of the audio stream of the audio device based on the orientation information specifically includes the following steps: The basic phase offset is calculated based on the actual straight-line distance from the azimuth information. Calculate the first azimuth compensation factor based on the azimuth and elevation angles of the azimuth information; The final phase adjustment parameters are calculated based on the basic phase offset and the first azimuth compensation factor. The audio stream is phase-rotated based on the final phase adjustment parameters.

[0053] In this embodiment of the application, based on the actual straight-line distance Calculate the basic phase offset Among them, the basic phase offset Represented as:

[0054] in, This represents the actual straight-line distance between the audio device and the TV host. This represents the wavelength of the audio signal (e.g., 440Hz corresponds to λ≈0.78m).

[0055] In this embodiment of the application, a first azimuth compensation factor is calculated based on the azimuth angle and the elevation angle, wherein the first azimuth compensation factor... Represented as:

[0056] in, Indicates azimuth. Indicates pitch angle, , This represents the empirical coefficient.

[0057] In this embodiment of the application, based on the basic phase offset and the first-position compensation factor Calculate the final phase adjustment parameters, where the final phase adjustment parameters are... Represented as:

[0058] In this embodiment of the application, a phase rotation operation is performed on the audio stream according to the final phase adjustment parameters to obtain an adjusted audio stream, wherein the adjusted audio stream... Represented as:

[0059] in, Represents an audio stream. This indicates the center frequency of the audio signal.

[0060] In practical applications, multiple audio devices (such as left and right channel headphones and surround sound speakers) need to receive wireless signals sent by the TV host. Due to different signal transmission paths (such as distance and obstacles), the signals received by each device may have a phase difference, resulting in audio-visual asynchrony or distorted stereo effect. In this case, phase rotation is required to calibrate the phase difference. Specifically... (1) The TV host transmits a test signal (such as a sine wave) and records the phase angle (such as 0°) at the time of transmission. (2) After each audio device receives the signal, measure the actual received phase angle (e.g., 30° for the left earphone and -15° for the right earphone). (3) Calculate the final phase adjustment parameters: The final phase adjustment parameter for the left earphone is 30° - 0° = 30°. The final phase adjustment parameter for the right earphone is -15° - 0° = -15°. (4) In the digital signal processor (DSP) of the audio device, the received signal is phase-rotated: • Left earphone signal: Rotate -30° (i.e., rotate 30° counterclockwise) to align the phase and transmit the signal; • Right earphone signal: Rotate 15° (i.e., rotate 15° clockwise) to eliminate phase difference.

[0061] Compared with existing technologies, this application achieves fully automated and high-precision dynamic phase calibration by integrating orientation information and audio stream.

[0062] In some optional implementations of this application, the step of performing delay calibration processing on the delay parameters of the audio stream based on the orientation information specifically includes the following steps: The base delay is calculated based on the actual straight-line distance from the azimuth information. Calculate the second azimuth compensation factor based on the azimuth and elevation angles of the azimuth information; The final delay adjustment parameters are calculated based on the base delay amount and the second-position compensation factor. The audio stream is subjected to delay compensation based on the final delay adjustment parameters.

[0063] In this embodiment of the application, based on the actual straight-line distance Calculate the base delay Among them, the basic delay amount Represented as:

[0064] in, Indicates the speed of sound propagation ( 343m / s The image delay parameter of an audio device can be obtained by playing a test signal and recording the arrival time of the image using professional testing tools (such as the REW audio analyzer) or a mobile app (such as Audio Delay Test).

[0065] In this embodiment of the application, based on the azimuth angle and pitch angle Calculate the second-azimuth compensation factor, where, the second-azimuth compensation factor Represented as: .

[0066] In this embodiment of the application, based on the basic delay amount and the second-direction compensation factor Calculate the final delay adjustment parameters. , Represents the empirical coefficient, where the final delay adjustment parameter is... Represented as: .

[0067] In this embodiment, a delay compensation operation is performed on the audio stream according to the final delay adjustment parameters to obtain a delay-compensated audio stream. Represented as: .

[0068] in, Represents an audio stream. Indicates audio stream The audio stream after delay compensation.

[0069] In practical applications, assuming the measurement results are: video delay 0ms and audio delay 100ms; therefore, it is necessary to play the audio 100ms earlier or delay the video by 100ms (usually, the audio is adjusted because video delay may affect the interactive experience). Then, the compensation value = target delay - actual delay = 0ms - 100ms = -100ms (i.e., the audio is played 100ms earlier). At this time, this application can adjust the "audio delay" parameter to -100ms in the speaker settings to ensure that the signals from different paths arrive synchronously.

[0070] Compared with existing technologies, this application proposes a fully automated, high-precision dynamic delay calibration method by fusing orientation information and audio stream, achieving millisecond-level response and sub-millimeter-level accuracy.

[0071] In some optional implementations of this application, the step of performing channel calibration processing on the channel allocation of the audio stream based on the orientation information specifically includes the following steps: A channel allocation weight matrix is ​​constructed based on psychoacoustic principles, wherein the channel allocation weight matrix includes left channel weight, center channel weight, right channel weight, surround left channel weight, and surround right channel weight. The azimuth angle of the azimuth information is mapped to the left channel weight, the middle channel weight, and the right channel weight; Construct a pitch angle attenuation factor based on the pitch angle of the azimuth information; The pitch attenuation factor is mapped to the left surround channel weight and the right surround channel weight to obtain the mapped channel allocation weight matrix. The audio stream is mixed using the mapped channel weighting matrix.

[0072] In this embodiment, a channel allocation weight matrix is ​​constructed based on psychoacoustic principles. Psychoacoustic principles refer to the mathematical description of the statistical properties of human hearing, explaining the physiological principles behind various auditory sensations. The aforementioned channel allocation weight matrix W is expressed as:

[0073] in, Indicates the first The energy distribution coefficient of the vocal tract ( ); Mapping the azimuth angle to the left, middle, and right channel weights in the channel weighting matrix yields the mapped left channel weight. Mid-channel weight and right channel weight ,in:

[0074]

[0075]

[0076] A pitch attenuation factor is constructed based on the pitch angle, where the pitch attenuation factor is... Represented as:

[0077] Mapping the pitch attenuation factor to the surround left channel weight and surround right channel weight in the channel assignment weight matrix yields the mapped surround left channel weight. and surround right channel weight ,in:

[0078]

[0079] The audio stream is mixed using the mapped channel weighting matrix to obtain the multichannel mixed audio stream. ,in:

[0080] in, Represents an audio stream. Indicates audio stream The audio stream after delay compensation. This indicates the vocal channel index (e.g., L / C / R / SL / SR).

[0081] Compared with existing technologies, this application realizes the direct conversion of orientation information to channel allocation scheme through mathematical modeling, which solves the problems of traditional methods relying on fixed layout and inaccurate sound image positioning, and significantly improves the immersion and user experience of multi-channel audio systems.

[0082] In some optional implementations of this application, the audio device includes a subwoofer device, and the step of performing audio calibration processing on the audio stream of the audio device based on the orientation information specifically includes the following steps: Determine if the subwoofer is located inside a cabinet; If the subwoofer is located in a cabinet, adjust the low-frequency gain of the audio stream; If the subwoofer is not located in a cabinet, do not adjust the low-frequency gain of the audio stream.

[0083] In this embodiment of the application, low-frequency energy analysis technology can be used to determine whether the subwoofer device is located in a cabinet. Specifically: (1) Transmit a 20-200Hz sweep frequency signal and record the direct sound energy. With reflected sound energy ; (2) Calculate the reflected energy ratio R= ; (3) If R> If so, the subwoofer is located inside the cabinet. This represents a pre-set energy threshold; for example, this energy threshold could be 1.5.

[0084] In this embodiment, when the subwoofer device is detected to be located inside a cabinet, a cabinet compensation filter can be applied. Adjust the low-frequency gain of the audio stream, specifically:

[0085] in, Indicates the attenuation magnitude. Indicates bandwidth control. This indicates the resonant center frequency.

[0086] Compared with existing technologies, this application proposes a fully automated, environment-adaptive subwoofer calibration method by integrating orientation perception and environmental recognition technologies, thereby achieving precise low-frequency control and seamless sound-image fusion.

[0087] In some optional implementations of the embodiments of this application, the step of determining whether the subwoofer device is located in the cabinet specifically includes the following steps: The frequency domain characteristics are obtained by collecting vibration data of the subwoofer device using an accelerometer. Determine whether the vibration energy increase of the frequency domain characteristics within a preset frequency band is greater than a preset increase threshold; If the vibration energy increase is greater than the preset increase threshold, the subwoofer device is determined to be located in the cabinet; otherwise, the subwoofer device is determined to be located outside the cabinet.

[0088] In this embodiment, the subwoofer vibration signal a(t) can be acquired using a triaxial accelerometer, with a sampling frequency of... =1kHz, duration T=100ms; then, a Butterworth low-pass filter (cutoff frequency) is applied to the original signal. =200Hz), suppressing high-frequency noise:

[0089] Finally, the filtered signal is divided into N segments according to the time window (e.g., N=10, each segment is 10ms) for subsequent frequency domain analysis.

[0090] In this embodiment of the application, after acquiring the filtered signal, each signal segment is... Perform a Fast Fourier Transform (FFT) to obtain the frequency domain representation. Then, calculate the vibration energy in the preset frequency band (e.g., 50-100Hz). :

[0091] Finally, this application averages the energy of the N signal segments to obtain the vibration energy characteristics at the current moment. :

[0092] In this embodiment of the application, a vibration energy amplification threshold Δ is predefined. Its value is determined based on: In open environments, the vibration energy of a subwoofer in the 50-100Hz frequency range is mainly generated by the vibration of the device itself, with a relatively small energy increase. • In a cabinet environment, the enclosed nature of the cabinet leads to enhanced low-frequency reflections, and the vibration energy increases significantly in the 50-100Hz frequency band (increase ≥30%). • Experimental calibration: In a standard cabinet (dimensions 1m × 0.5m × 0.5m, with good sealing), Δ The value is taken as the base energy. 1.3 times (i.e., Δ) =1.3 ).

[0093] In this embodiment of the application, the vibration energy amplification threshold Δ is defined. Next, this application calculates the ratio R between the current vibration energy and the reference energy:

[0094] in, The average vibration energy of the device in an open environment (which can be obtained through initial calibration or historical data statistics).

[0095] In this embodiment of the application, if the ratio R is greater than the vibration energy amplification threshold Δ If the subwoofer is located inside the cabinet, then the ratio R is less than or equal to the vibration energy amplification threshold Δ. If so, the subwoofer is determined to be located in an open environment.

[0096] Compared with existing technologies, this application achieves accurate perception of environmental conditions by analyzing the vibration characteristics of the subwoofer itself, thus solving the problems of low detection accuracy and poor real-time performance in existing technologies.

[0097] The embodiments of this application can acquire and process relevant data based on artificial intelligence technology. Artificial intelligence (AI) refers to the theories, methods, technologies, and application systems that use digital computers or machines controlled by digital computers to simulate, extend, and expand human intelligence, perceive the environment, acquire knowledge, and use that knowledge to obtain optimal results.

[0098] Foundational technologies for artificial intelligence generally include sensors, dedicated AI chips, cloud computing, distributed storage, big data processing, operating / interactive systems, and mechatronics. AI software technologies mainly encompass computer vision, robotics, biometrics, speech processing, natural language processing, and machine learning / deep learning.

[0099] Those skilled in the art will understand that all or part of the processes in the methods of the above embodiments can be implemented by instructing related hardware through computer-readable instructions. These computer-readable instructions can be stored in a computer-readable storage medium. When the program is executed, it can include the processes of the embodiments of the above methods. The aforementioned storage medium can be a non-volatile storage medium such as a magnetic disk, optical disk, or read-only memory (ROM), or random access memory (RAM).

[0100] It should be understood that although the steps in the flowcharts of the accompanying figures are shown sequentially as indicated by the arrows, these steps are not necessarily executed in the order indicated by the arrows. Unless explicitly stated herein, there is no strict order restriction on the execution of these steps, and they can be executed in other orders. Moreover, at least some steps in the flowcharts of the accompanying figures may include multiple sub-steps or multiple stages. These sub-steps or stages are not necessarily completed at the same time, but can be executed at different times, and their execution order is not necessarily sequential, but can be performed alternately or in turn with other steps or at least some of the sub-steps or stages of other steps.

[0101] Further reference Figure 3 As a response to the above Figure 2 The implementation of the method shown in this application provides an embodiment of an audio calibration device applied to a distributed television audio system. This device embodiment is similar to... Figure 2 Corresponding to the method embodiments shown, this device can be specifically applied to various electronic devices.

[0102] like Figure 3 As shown, the audio calibration device 200 for a distributed television audio system according to an embodiment of this application includes: The spatial topology recognition module 210 is used to perform spatial topology recognition operations on the TV host and several audio devices placed in the target space to obtain three-dimensional spatial data of the audio devices relative to the TV host. The orientation information calculation module 220 is used to calculate the orientation information of the audio device based on the three-dimensional spatial data. Audio calibration module 230 is used to perform audio calibration processing on the audio stream of the audio device according to the orientation information; The audio calibration module 230 includes: a phase calibration submodule 231, a delay calibration submodule 232, and a channel calibration submodule 233, wherein: The phase calibration submodule 231 is used to perform phase calibration processing on the phase parameters of the audio stream based on the orientation information; The delay calibration submodule 232 is used to perform delay calibration processing on the delay parameters of the audio stream according to the orientation information; The channel calibration submodule 233 is used to perform channel calibration processing on the channel allocation of the audio stream according to the orientation information.

[0103] In this embodiment of the application, an audio calibration device 200 for a distributed television audio system is provided, comprising: a spatial topology identification module 210, used to perform spatial topology identification operations on a television host and several audio devices placed in a target space to obtain three-dimensional spatial data of the audio devices relative to the television host; a directional information calculation module 220, used to calculate the directional information of the audio devices based on the three-dimensional spatial data; and an audio calibration module 230, used to perform audio calibration processing on the audio stream of the audio devices based on the directional information. The audio calibration module 230 includes: a phase calibration submodule 231, a delay calibration submodule 232, and a channel calibration submodule 233, wherein: the phase calibration submodule 231 is used to perform phase calibration processing on the phase parameters of the audio stream based on the directional information; the delay calibration submodule 232 is used to perform delay calibration processing on the delay parameters of the audio stream based on the directional information; and the channel calibration submodule 233 is used to perform channel calibration processing on the channel allocation of the audio stream based on the directional information. Compared with existing technologies, this application can accurately identify the position information of the audio device relative to the TV host, and accurately calibrate the audio stream of the audio device based on this information, thereby improving the audio playback quality of the distributed TV audio system and providing users with a better and more realistic listening experience.

[0104] In some optional implementations of the embodiments of this application, the television host includes a signal transmitter, the audio device includes a signal receiver, and the aforementioned spatial topology identification module includes: The signal transmission submodule is used to transmit low-frequency detection signals to the target space according to the signal transmitter. The signal receiving submodule is used to obtain the signal arrival time and signal arrival strength based on the low-frequency detection signal received by the signal receiver. The 3D data calculation submodule is used to calculate the 3D spatial data of the audio device relative to the TV host based on the signal arrival time and signal strength.

[0105] In some optional implementations of the embodiments of this application, the above-mentioned three-dimensional data calculation submodule includes: The signal propagation time calculation unit is used to calculate the signal propagation time based on the transmission time and arrival time of the low-frequency detection signal. The straight-line distance calculation unit is used to calculate the predicted straight-line distance between the audio equipment and the television host based on the signal propagation time. The signal strength calculation unit is used to construct and predict the signal strength corresponding to the straight-line distance based on the model of signal strength attenuation with distance; An auxiliary distance acquisition unit is used to convert signal intensity into auxiliary distance based on the fitted model; The weighted fusion unit is used to weight and fuse the predicted straight-line distance based on the auxiliary distance to obtain the true straight-line distance. The three-dimensional data calculation unit is used to calculate the three-dimensional spatial data of the audio device relative to the TV host based on the actual straight-line distance and the trilateration method.

[0106] In some optional implementations of the embodiments of this application, the phase calibration submodule includes: The basic phase offset calculation unit is used to calculate the basic phase offset based on the true straight-line distance of the azimuth information. The first azimuth compensation factor calculation unit is used to calculate the first azimuth compensation factor based on the azimuth angle and elevation angle of the azimuth information. The final phase adjustment parameter calculation unit is used to calculate the final phase adjustment parameters based on the basic phase offset and the first azimuth compensation factor. The phase rotation operation unit is used to perform phase rotation operation on the audio stream according to the final phase adjustment parameters.

[0107] In some optional implementations of the embodiments of this application, the delay calibration submodule includes: The basic delay calculation unit is used to calculate the basic delay based on the actual straight-line distance of the azimuth information. The second azimuth compensation factor calculation unit is used to calculate the second azimuth compensation factor based on the azimuth angle and elevation angle of the azimuth information. The final delay adjustment parameter calculation unit is used to calculate the final delay adjustment parameters based on the basic delay amount and the second orientation compensation factor. The delay compensation operation unit is used to perform delay compensation operation on the audio stream according to the final delay adjustment parameters.

[0108] In some optional implementations of this application's embodiments, the audio channel calibration submodule includes: The channel allocation weight matrix construction unit is used to construct a channel allocation weight matrix based on psychoacoustic principles. The channel allocation weight matrix includes left channel weight, center channel weight, right channel weight, surround left channel weight, and surround right channel weight. An azimuth mapping unit is used to map the azimuth of the azimuth information to the left channel weight, the middle channel weight, and the right channel weight. A pitch angle attenuation factor construction unit is used to construct a pitch angle attenuation factor based on the pitch angle of the azimuth information. Pitch angle attenuation factor mapping unit is used to map the pitch angle attenuation factor to the surround left channel weight and the surround right channel weight to obtain the mapped channel allocation weight matrix. A multi-channel mixing operation unit is used to perform multi-channel mixing operation on the audio stream according to the mapped channel allocation weight matrix.

[0109] To address the aforementioned technical problems, embodiments of this application also provide a computer device. Please refer to [link / reference needed]. Figure 4 , Figure 4 This is a basic structural block diagram of a computer device according to an embodiment of this application.

[0110] Computer device 300 includes a memory 310, a processor 320, and a network interface 330 that are interconnected via a system bus. It should be noted that only computer device 300 with components 310-330 is shown in the figure; however, it should be understood that it is not required to implement all the shown components, and more or fewer components can be implemented alternatively. Those skilled in the art will understand that the computer device described here is a device capable of automatically performing numerical calculations and / or information processing according to pre-set or stored instructions, and its hardware includes, but is not limited to, microprocessors, application-specific integrated circuits (ASICs), field-programmable gate arrays (FPGAs), digital signal processors (DSPs), embedded devices, etc.

[0111] Computer devices can include desktop computers, laptops, handheld computers, and cloud servers. These devices allow for human-computer interaction with users through keyboards, mice, remote controls, touchpads, or voice-activated devices.

[0112] The memory 310 includes at least one type of readable storage medium, including flash memory, hard disk, multimedia card, card-type memory (e.g., SD or DX memory), random access memory (RAM), static random access memory (SRAM), read-only memory (ROM), electrically erasable programmable read-only memory (EEPROM), programmable read-only memory (PROM), magnetic memory, magnetic disk, optical disk, etc. In some embodiments, the memory 310 may be an internal storage unit of the computer device 300, such as the hard disk or memory of the computer device 300. In other embodiments, the memory 310 may also be an external storage device of the computer device 300, such as a plug-in hard disk, smart media card (SMC), secure digital (SD) card, flash card, etc., equipped on the computer device 300. Of course, the memory 310 may include both internal storage units and external storage devices of the computer device 300. In the embodiments of this application, the memory 310 is typically used to store the operating system and various application software installed on the computer device 300, such as computer-readable instructions for audio calibration methods applied to a distributed television audio system. In addition, the memory 310 can also be used to temporarily store various types of data that have been output or will be output.

[0113] In some embodiments, processor 320 may be a central processing unit (CPU), controller, microcontroller, microprocessor, or other data processing chip. Processor 320 is typically used to control the overall operation of computer device 300. In embodiments of this application, processor 320 is used to execute computer-readable instructions stored in memory 310 or process data, such as executing computer-readable instructions for an audio calibration method applied to a distributed television audio system.

[0114] The network interface 330 may include a wireless network interface or a wired network interface, which is typically used to establish a communication connection between the computer device 300 and other electronic devices.

[0115] The computer equipment provided in this application can accurately identify the position information of the audio device relative to the TV host, and accurately calibrate the audio stream of the audio device based on this information, thereby improving the audio playback quality of the distributed TV audio system and providing users with a better and more realistic listening experience.

[0116] This application also provides another embodiment, namely, a computer-readable storage medium storing computer-readable instructions that can be executed by at least one processor to cause the at least one processor to perform the steps of the audio calibration method applied to a distributed television audio system as described above.

[0117] The computer-readable storage medium provided in this application can accurately identify the position information of the audio device relative to the TV host, and accurately calibrate the audio stream of the audio device based on this information, thereby improving the audio playback quality of the distributed TV audio system and providing users with a better and more realistic listening experience.

[0118] Through the above description of the embodiments, those skilled in the art can clearly understand that the methods of the above embodiments can be implemented by means of software plus necessary general-purpose hardware platforms. Of course, they can also be implemented by hardware, but in many cases the former is a better implementation method. Based on this understanding, the technical solution of this application, in essence, or the part that contributes to the prior art, can be embodied in the form of a software product. This computer software product is stored in a storage medium (such as ROM / RAM, magnetic disk, optical disk) and includes several instructions to cause a terminal device (which may be a mobile phone, computer, server, air conditioner, or network device, etc.) to execute the methods of the various embodiments of this application.

[0119] Obviously, the embodiments described above are only some embodiments of this application, not all embodiments. The accompanying drawings show preferred embodiments of this application, but do not limit the patent scope of this application. This application can be implemented in many different forms; rather, the purpose of providing these embodiments is to provide a more thorough and comprehensive understanding of the disclosure of this application. Although this application has been described in detail with reference to the foregoing embodiments, those skilled in the art can still modify the technical solutions described in the foregoing specific embodiments, or make equivalent substitutions for some of the technical features. Any equivalent structures made using the content of this application's specification and drawings, directly or indirectly applied to other related technical fields, are similarly within the scope of patent protection of this application.

Claims

1. An audio calibration method for a distributed television audio system, characterized in that, Includes the following steps: A spatial topology recognition operation is performed on the TV host and several audio devices placed in the target space to obtain the three-dimensional spatial data of the audio devices relative to the TV host. Calculate the orientation information of the audio device based on the three-dimensional spatial data; The audio stream of the audio device is calibrated based on the location information. The step of performing audio calibration processing on the audio stream of the audio device based on the orientation information specifically includes the following steps: The phase parameters of the audio stream are calibrated based on the azimuth information. The delay parameters of the audio stream are calibrated based on the location information. Based on the location information, the channel allocation of the audio stream is calibrated.

2. The audio calibration method for a distributed television audio system according to claim 1, characterized in that, The television host includes a signal transmitter, and the audio device includes a signal receiver. The step of performing spatial topology recognition on the television host and several audio devices placed in the target space to obtain three-dimensional spatial data of the audio devices relative to the television host specifically includes the following steps: The signal transmitter sends a low-frequency detection signal to the target space. Based on the low-frequency detection signal received by the signal receiver, the signal arrival time and signal arrival strength are obtained; The three-dimensional spatial data of the audio device relative to the television host are calculated based on the signal arrival time and the signal arrival strength.

3. The audio calibration method for a distributed television audio system according to claim 2, characterized in that, The step of calculating the three-dimensional spatial data of the audio device relative to the television host based on the signal arrival time and the signal arrival strength specifically includes the following steps: The signal propagation time is calculated based on the transmission time of the low-frequency detection signal and the arrival time of the signal. Calculate the predicted straight-line distance between the audio device and the television host based on the signal propagation time; The signal strength corresponding to the predicted straight-line distance is constructed based on the model of signal strength attenuation with distance; The signal strength is converted into an auxiliary distance based on the fitted model; The predicted straight-line distance is weighted and fused based on the auxiliary distance to obtain the true straight-line distance; The three-dimensional spatial data of the audio device relative to the television host are calculated based on the actual straight-line distance and the trilateration method.

4. The audio calibration method for a distributed television audio system according to claim 1, characterized in that, The step of performing phase calibration processing on the phase parameters of the audio stream of the audio device based on the orientation information specifically includes the following steps: The basic phase offset is calculated based on the actual straight-line distance of the aforementioned azimuth information; Calculate the first azimuth compensation factor based on the azimuth and elevation angles of the azimuth information; The final phase adjustment parameters are calculated based on the basic phase offset and the first azimuth compensation factor. The audio stream is phase-rotated according to the final phase adjustment parameters.

5. The audio calibration method for a distributed television audio system according to claim 1, characterized in that, The step of performing delay calibration processing on the delay parameters of the audio stream based on the orientation information specifically includes the following steps: The base delay is calculated based on the actual straight-line distance of the aforementioned orientation information; Calculate the second azimuth compensation factor based on the azimuth and elevation angles of the azimuth information; The final delay adjustment parameters are calculated based on the basic delay amount and the second orientation compensation factor. The audio stream is subjected to delay compensation operation based on the final delay adjustment parameters.

6. The audio calibration method for a distributed television audio system according to claim 1, characterized in that, The step of performing channel calibration processing on the channel allocation of the audio stream based on the orientation information specifically includes the following steps: A channel allocation weight matrix is ​​constructed based on psychoacoustic principles, wherein the channel allocation weight matrix includes left channel weight, center channel weight, right channel weight, surround left channel weight, and surround right channel weight. The azimuth angle of the azimuth information is mapped to the left channel weight, the middle channel weight, and the right channel weight; Construct a pitch angle attenuation factor based on the pitch angle of the azimuth information; The pitch attenuation factor is mapped to the left surround channel weight and the right surround channel weight to obtain the mapped channel allocation weight matrix. The audio stream is mixed using the mapped channel weighting matrix.

7. The audio calibration method for a distributed television audio system according to claim 1, characterized in that, The audio device includes a subwoofer device, and the step of performing audio calibration processing on the audio stream of the audio device based on the orientation information specifically includes the following steps: Determine whether the subwoofer device is located inside the cabinet; If the subwoofer device is located in a cabinet, adjust the low-frequency gain of the audio stream parameters; If the subwoofer device is not located in the cabinet, the low-frequency gain of the audio stream parameters is not adjusted.

8. An audio calibration device for use in a distributed television audio system, characterized in that, include: The spatial topology recognition module is used to perform spatial topology recognition on a TV host and several audio devices placed in the target space, and obtain three-dimensional spatial data of the audio devices relative to the TV host. A location information calculation module is used to calculate the location information of the audio device based on the three-dimensional spatial data. An audio calibration module is used to perform audio calibration processing on the audio stream of the audio device based on the orientation information; The audio calibration module includes: a phase calibration submodule, a delay calibration submodule, and a channel calibration submodule, wherein: The phase calibration submodule is used to perform phase calibration processing on the phase parameters of the audio stream based on the orientation information; The delay calibration submodule is used to perform delay calibration processing on the delay parameters of the audio stream based on the orientation information; The channel calibration submodule is used to perform channel calibration processing on the channel allocation of the audio stream based on the orientation information.

9. A computer device, comprising a memory and a processor, characterized in that, The memory stores computer-readable instructions, and when the processor executes the computer-readable instructions, it implements the steps of the audio calibration method for a distributed television audio system as described in any one of claims 1 to 7.

10. A computer-readable storage medium, characterized in that, The computer-readable storage medium stores computer-readable instructions, which, when executed by a processor, implement the steps of the audio calibration method for a distributed television audio system as described in any one of claims 1 to 7.

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