A portable multi-dimensional surround sound playback system

Abstract

The application discloses a kind of portable multi-dimensional surround sound playback system, it is related to audio playback technical field.The system includes: main sound box and multiple satellite sound box, audio processing module, wireless transmitting module and power management module are integrated in main sound box, wireless receiving module, power amplifier module and full-frequency loudspeaker unit are integrated in each satellite sound box.Audio processing module receives original multi-channel audio data and based on sound field spatial feature model, it is disassembled into direct sound field data stream and scattered sound field data stream;Wireless transmitting module packs direct sound field data stream into first data packet sequence, and packs scattered sound field data stream into second data packet sequence;Wireless receiving module extracts corresponding channel data from first data packet sequence according to the real-time orientation information of satellite sound box relative to main sound box, and extracts environmental filling data from second data packet sequence.

Description

Technical Field

[0001] This invention relates to the field of audio playback technology, specifically a portable multi-dimensional surround audio playback system. Background Technology

[0002] Portable surround sound playback devices are widely used in mobile entertainment, outdoor audio-visual scenarios, and other applications. Their core requirement is to reproduce a multi-dimensional sound field within a limited size and under wireless transmission conditions. Existing portable surround systems typically employ a fixed channel allocation scheme. The main speaker wirelessly transmits the audio signals of each surround channel to satellite speakers with pre-defined channel roles. Each satellite speaker then replays sound content from a specific direction. This scheme requires the satellite speakers to be placed strictly according to the preset orientation. If the user cannot place the satellite speakers in the standard position due to environmental constraints, or if the satellite speakers are moved during use, it will cause sound image localization errors. The direction of the sound source perceived by the listener will deviate from the actual or expected direction, significantly reducing the surround sound and immersion. Furthermore, traditional portable surround systems often use a uniform packetization method for wireless transmission of audio data, directly encapsulating the data streams of each channel into data packets with the same priority and synchronization characteristics for transmission. This approach fails to distinguish between directional and diffuse environmental components in the audio content. This makes it difficult to flexibly extract the direct sound components crucial for sound image localization and the ambient sound components used to create a sense of immersion when wireless link quality fluctuates or the receiver needs to adjust the playback content based on the actual location. When the signal is interfered with or the sound field needs to be reconstructed based on the real-time location of the satellite speakers, conventional solutions lack a mechanism to decompose sound field information into independently adjustable components, affecting the accuracy and flexibility of sound field reconstruction. Therefore, maintaining accurate sound image localization and a stable sense of immersion in a multi-dimensional surround sound field under portable usage conditions where the satellite speaker placement is variable is a current challenge for portable surround audio systems. Summary of the Invention

[0003] The purpose of this invention is to provide a portable multi-dimensional surround audio playback system that can accurately reproduce the sound source directionality and spatial immersion of a multi-channel surround sound field even when the position of the satellite speakers relative to the main speakers can be arbitrarily changed.

[0004] To achieve the above objectives, the present invention provides the following technical solution: a portable multi-dimensional surround audio playback system, comprising a main speaker enclosure, multiple satellite speaker enclosures, and an optional mobile control terminal. An audio processing module, a wireless transmission module, and a power management module are integrated inside the main speaker enclosure. Each satellite speaker enclosure integrates a wireless receiving module, a power amplification module, and a full-range speaker unit.

[0005] The audio processing module receives raw multi-channel audio data and, based on a built-in sound field spatial feature model, decomposes the raw multi-channel audio data into a direct sound field data stream reflecting the directional characteristics of sound and a scattered sound field data stream reflecting the environmental envelopment characteristics. As a technical solution of this invention, the audio processing module performs frame segmentation on the raw multi-channel audio data to obtain multiple audio frame data blocks, and performs a time-frequency transformation on each audio frame data block to obtain a frequency domain representation matrix. Preferably, the time-frequency transformation uses a modified discrete cosine transform to map the audio frame data block from the time domain to the frequency domain, outputting a frequency domain coefficient matrix as the frequency domain representation matrix. The audio processing module performs spatial sparse decomposition on the frequency domain representation matrix according to a predefined direction vector dictionary in the sound field spatial feature model, separating the directional sound field components and the environmental sound field components. Then, the time-domain reconstructed signal of the directional sound field components is labeled as a direct sound field data stream, and the time-domain reconstructed signal of the environmental sound field components is labeled as a scattered sound field data stream. By disassembling the sound field in this way, the system can accurately reproduce the direct sound with clear spatial directionality and the scattered sound that creates spatial atmosphere from the original multi-channel material, laying a data foundation for the subsequent flexible reconstruction of the sound field based on the actual speaker placement.

[0006] The wireless transmission module packages the direct sound field data stream into a first data packet sequence and the scattered sound field data stream into a second data packet sequence. As a technical solution of this invention, the wireless transmission module acquires an interference frequency distribution map in the current network environment and, based on this, preferentially selects the frequency hopping sequence with the fewest intersections with interference frequencies from a preset frequency hopping spectrum as the current frequency hopping sequence. Using the current frequency hopping sequence, the direct sound field data stream is segmented and encapsulated into multiple first data frames, and a first synchronization timestamp is inserted into the frame header to obtain the first data packet sequence. Simultaneously, the scattered sound field data stream is segmented and encapsulated into multiple second data frames, and a second synchronization timestamp is inserted into the frame header to obtain the second data packet sequence, wherein the effective payload length of the first data frame is greater than the effective payload length of the second data frame. This differentiated packaging and synchronous transmission strategy ensures higher data integrity for the direct sound field data while fully utilizing wireless channel resources, effectively avoiding interference frequencies, and ensuring low latency and high stability when transmitting multiple audio data streams in the air.

[0007] Wireless receiving modules, located within satellite speaker enclosures at various locations, receive a first data packet sequence and a second data packet sequence, and perform synchronous demodulation to obtain a first synchronous data stream and a second synchronous data stream. The wireless receiving module reads attitude angle data output from the inertial measurement unit built into the satellite speaker enclosure and calculates the real-time azimuth information of the satellite speaker enclosure relative to the main speaker enclosure based on this attitude angle data. According to this real-time azimuth information, it retrieves the target channel identifier corresponding to the current physical location of the satellite speaker enclosure from a preset channel mapping table. Subsequently, the wireless receiving module filters audio data packets carrying the target channel identifier from the first synchronous data stream, extracting their internal payload as channel data; simultaneously, it performs mean filtering on all data packets in the second synchronous data stream to obtain environmental filling data. Preferably, this mean filtering uses an energy-weighted averaging method, allocating weight coefficients according to the spatial diffusion coefficient carried by each data packet, thereby generating more spatially consistent environmental filling data. This solution enables each satellite speaker to dynamically analyze its actual position in space, based on its random placement, to determine the directional channel signal that should be emitted from that position, and to superimpose and integrate environmental information, thus truly achieving adaptive sound field reconstruction within the listening area.

[0008] The power amplification module merges the resolved channel data and ambient fill data into a driving signal to drive the full-range speaker unit. Specifically, a large first gain coefficient is applied to the channel data to obtain a first amplified signal, and a smaller second gain coefficient is applied to the ambient fill data to obtain a second amplified signal, ensuring the clarity and dominance of the directional sound source. The first and second amplified signals are time-domain superimposed to generate a merged analog signal, which is then converted into a driving signal and output to the voice coil of the full-range speaker unit using pulse width modulation technology. This merged driving method ensures accurate and sharp sound positioning while filling the listening space with ambient sound of lower energy dispersion, creating a natural and immersive multi-dimensional surround sound field.

[0009] As a preferred embodiment of the present invention, the system further includes a mobile control terminal. This mobile control terminal acquires two-dimensional planar map data of the target listening area and marks the main speaker's primary position coordinates on it. Using the primary position coordinates as the center and a preset multi-dimensional surround sound field coverage radius as the reference radius, the mobile control terminal generates an initial circular layout line on the two-dimensional planar map. Collision detection is performed based on obstacle outlines in the map data to obtain a set of layable arc segments. Based on the total number of satellite speakers, multiple initial candidate points are sampled at equal angular intervals on the set of layable arc segments. Random jitter offsets are applied to these initial candidate points, ultimately generating a sequence of candidate satellite speaker positions that conforms to the actual room environment and is easy for users to operate. Satellite speaker layout guidance information is then generated accordingly. This significantly reduces the difficulty for users to build a multi-channel surround system, enabling rapid and reasonable speaker layout even in non-professional home environments.

[0010] The power management module includes a battery power prediction submodule. This submodule collects the battery terminal voltage and current of the main speaker in real time, calculates the current remaining battery percentage, and then looks up the matching transmission power level from a preset power level lookup table based on this percentage. It then generates a power control command to adjust the transmission power of the radio frequency front-end circuit of the wireless transmission module. This closed-loop power regulation mechanism intelligently reduces radio frequency power consumption when the battery power decreases, extending the system's usable battery life in mobile scenarios.

[0011] Furthermore, the audio processing module also has the function of dynamically adjusting the energy allocation ratio of the direct sound field data stream and the scattered sound field data stream. The audio processing module performs rhythmic feature analysis on the raw multi-channel audio data to obtain the beat density value of the current audio segment. When the beat density value is greater than a preset high-density threshold, it indicates that the audio content is rhythmically intense and requires a strong sense of direction. The audio processing module adjusts the energy allocation ratio to a first allocation ratio with a larger proportion of energy from the direct sound field data stream. When the beat density value is less than a preset low-density threshold, it indicates that the audio content is soothing and requires a better sense of spatial immersion. In this case, the energy allocation ratio is adjusted to a second allocation ratio with a larger proportion of energy from the scattered sound field data stream. This adaptive processing allows the system to automatically optimize the sound field rendering emphasis based on the emotional and rhythmic changes of the music content itself, maintaining the best listening experience.

[0012] As a further improvement of this invention, each satellite speaker enclosure is equipped with the ability to automatically compensate for sound pressure based on the listening distance. The distance sensing submodule within the satellite speaker enclosure emits an ultrasonic ranging signal and receives the reflected echo, calculating the distance between the satellite speaker enclosure and the listening position in real time based on the time difference between transmission and reception. The power amplification module searches for the corresponding compensation gain value in a pre-stored distance-sound pressure compensation curve based on this real-time distance, and multiplies the amplitude of the drive signal by this compensation gain value to obtain the final drive signal to drive the full-range speaker unit. This solution ensures a balanced and consistent loudness experience regardless of the distance the user places the satellite speaker at from the listening position, further enhancing the system's portability and environmental adaptability.

[0013] The technical effects and advantages provided by the present invention in the above technical solution are as follows:

[0014] The audio processing module, based on a sound field spatial characteristic model, decomposes the original multi-channel audio data into direct sound field data streams and scattered sound field data streams. The wireless transmission module then packages these two data streams into independent first and second data packet sequences, respectively. This decomposition and packaging method allows for the differentiation of directional signal components, which are crucial for sound image localization, from environmental signal components used to create a sense of space, along the wireless transmission link. This allows the receiver to acquire and utilize these two types of sound field information according to actual needs. Therefore, when the satellite speaker's placement changes, the system does not need to retransmit all channel signals; it only needs to extract the directional components corresponding to the current orientation from the decomposed data streams, while flexibly incorporating environmental fill components to provide an information structure consistent with the actual acoustic propagation process for sound field reconstruction. The wireless receiving module, based on the real-time orientation information of the satellite speaker relative to the main speaker, extracts the channel data corresponding to the real-time orientation information from the first data packet sequence and simultaneously extracts the environmental fill data from the second data packet sequence. The power amplification module then merges these two data packets into a driving signal. This process allows satellite speakers to dynamically select the channel signals to be reproduced based on their actual spatial location and independently adjust the amount of ambient sound filling, without the need to predefine fixed channel roles. When users freely arrange the satellite speaker positions according to their listening environment, the sound image positioning automatically adapts to the actual direction, maintaining a consistent sense of environmental immersion. This avoids sound field distortion caused by placement errors, enabling the portable surround audio playback system to output a clear and spatially continuous multi-dimensional sound field in various usage layouts. Attached Figure Description

[0015] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the drawings used in the embodiments will be briefly introduced below. Obviously, the drawings described below are only some embodiments recorded in this invention. For those skilled in the art, other drawings can be obtained based on these drawings.

[0016] Figure 1 This is a schematic diagram of the structure of a portable multi-dimensional surround sound audio playback system;

[0017] Figure 2 This is a flowchart of the audio processing module;

[0018] Figure 3 This is a schematic diagram of the structure of the wireless transmission module and the frequency hopping data processing flow;

[0019] Figure 4 This is a schematic diagram of the wireless receiver module structure of the satellite speaker enclosure;

[0020] Figure 5 This is a schematic diagram showing the signal processing of the power amplifier module and the layout of the satellite speaker;

[0021] Figure 6 This is the flowchart of the adaptive control of the speaker system. Detailed Implementation

[0022] To make the objectives, technical solutions, and advantages of the embodiments of the present invention clearer, the technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0023] See Figure 1This invention provides a portable multi-dimensional surround sound playback system, comprising a main speaker enclosure and multiple satellite speaker enclosures. The main speaker enclosure integrates an audio processing module, a wireless transmission module, and a power management module. Each satellite speaker enclosure integrates a wireless receiving module, a power amplifier module, and a full-range speaker unit. During system operation, the audio processing module receives raw multi-channel audio data from an external source and decomposes the raw multi-channel audio data according to a built-in sound field spatial characteristic model, generating a direct sound field data stream and a scattered sound field data stream. These two data streams are transmitted to the wireless transmission module. The wireless transmission module packages the direct sound field data stream into a first data packet sequence and the scattered sound field data stream into a second data packet sequence, which are then broadcast via a wireless channel. Upon receiving the first and second data packet sequences, the wireless receiving modules in each satellite speaker enclosure extract data based on the real-time azimuth information of the satellite speaker enclosure relative to the main speaker enclosure. Specifically, they extract the channel data corresponding to the real-time azimuth information from the first data packet sequence and extract environmental fill data from the second data packet sequence. Then, the power amplification module merges the extracted channel data and ambient filling data to generate a drive signal, which ultimately drives the full-range speaker unit to produce sound.

[0024] Example 1:

[0025] In specific implementation, please refer to Figure 2 The audio processing module's processing of raw multi-channel audio data includes frame segmentation, time-frequency transformation, spatial sparse decomposition, and temporal reconstruction.

[0026] The audio processing module receives raw multi-channel audio data as a multi-channel pulse-code modulation (PCM) data stream. Internally, the audio processing module includes a frame segmentation unit. This unit performs windowing and framing operations on the raw multi-channel audio data with a preset frame length and frame overlap ratio, resulting in multiple audio frame data blocks. The preset frame length can be set to 512 or 1024 samples, and the frame overlap ratio is set to 50%. Each audio frame data block contains sampled data from all channels within the same time segment.

[0027] For each audio frame data block, the time-frequency transformation unit in the audio processing module uses a modified discrete cosine transform to map the audio frame data block from the time domain to the frequency domain. The transformation process of the modified discrete cosine transform is expressed as follows:

[0028]

[0029] in, Indicates the first The first audio frame data block The values ​​of each time-domain sampling point This represents the transform length of the modified discrete cosine transform. The value is equal to the frame length. For time-domain sampling point index, The range of values ​​is to integers, For frequency domain coefficient index, The range of values ​​is to integers, Indicates the first After transformation, the audio frame data block is at the... The frequency domain coefficients at each frequency point. The audio processing module performs the above-mentioned modified discrete cosine transform on each channel of each audio frame data block, and organizes the frequency domain coefficients of all channels at all frequency points into a frequency domain coefficient matrix. Each row of the frequency domain coefficient matrix corresponds to a frequency point, and each column corresponds to a channel. The frequency domain coefficient matrix is ​​the frequency domain representation matrix.

[0030] After obtaining the frequency domain representation matrix, the audio processing module calls the predefined direction vector dictionary in the sound field spatial feature model. The direction vector dictionary consists of a set of direction vectors, each corresponding to a spatial orientation. Each element in the direction vector represents the amplitude and phase response of the transfer function of an ideal sound source reaching each channel at a specific frequency point. The audio processing module uses the direction vector dictionary to perform spatial sparse decomposition on the frequency domain representation matrix. The process of spatial sparse decomposition can be described as follows: For each row vector in the frequency domain representation matrix, the audio processing module solves a sparse representation problem, such that the row vector can be approximated by a linear combination of a small number of direction vectors in the direction vector dictionary. The orientation of the direction vector corresponding to the non-zero coefficients in the combination coefficients is the directional sound field source, and the non-zero coefficients themselves constitute the coefficients of the directional sound field component at that frequency point. The part of the row vector that cannot be represented by the direction vector dictionary is attributed to the coefficients of the environmental sound field component at that frequency point. After traversing all frequency points, the audio processing module obtains the complete directional sound field component and the environmental sound field component.

[0031] The audio processing module performs temporal reconstruction on both the directional and ambient sound field components. The temporal reconstruction process is as follows: the audio processing module performs an inverse modified discrete cosine transform on the coefficients of the directional sound field component at each frequency point to obtain the corresponding temporal signal, and then combines the temporal signals of all audio frames into a continuous direct sound field data stream through overlapping and addition. Similarly, the audio processing module performs an inverse modified discrete cosine transform on the coefficients of the ambient sound field component at each frequency point to obtain the corresponding temporal signal, and then combines the temporal signals of all audio frames into a continuous scattered sound field data stream through overlapping and addition. The audio processing module outputs both the direct and scattered sound field data streams to the wireless transmission module.

[0032] Example 2:

[0033] In specific implementation, please refer to Figure 3 After acquiring the direct sound field data stream and scattered sound field data stream output by the audio processing module, the wireless transmission module performs data packetization and frequency hopping sequence selection operations before transmission. The wireless transmission module internally includes an interference sensing unit, a frequency hopping sequence selection unit, and a data packetization unit.

[0034] The interference sensing unit in the wireless transmission module is responsible for scanning the electromagnetic interference situation in the current network environment. The interference sensing unit acquires the received signal strength indication (RSI) values ​​of each sub-channel within the operating frequency band through broadband spectrum scanning, and records the center frequency of each sub-channel with its corresponding RSI value. The interference sensing unit marks sub-channels whose RSI values ​​exceed a preset interference threshold as interfered frequencies. The preset interference threshold can be set to -85dBm or dynamically adjusted to a value 15dB higher than the background noise floor based on the background noise floor of the wireless communication band in which the system is located. The set of center frequencies of all marked interfered frequencies constitutes the interference frequency distribution map in the current network environment.

[0035] The frequency hopping sequence selection unit obtains the interference frequency distribution map from the interference sensing unit and reads a preset frequency hopping spectrum from the internal memory of the wireless transmission module. The preset frequency hopping spectrum consists of several frequency hopping sequences, each corresponding to an ordered list of sub-channel center frequencies. This ordered arrangement specifies the order in which the radio frequency carrier frequency switches over time. The frequency hopping sequence selection unit calculates the number of intersections between each frequency hopping sequence and the interference frequencies in the interference frequency distribution map. The number of intersections is defined as the total number of sub-channel center frequencies appearing in the interference frequency distribution map for a given frequency hopping sequence. The selection algorithm used by the frequency hopping sequence selection unit is expressed as follows:

[0036]

[0037] in, This represents the preset frequency hopping pattern. For predefined A set of frequency-hopping sequences. This represents the total number of frequency hopping sequences. express The first in A frequency hopping sequence, This is the index number of the frequency hopping sequence. The range of values ​​is to Integers; This represents the distribution of interference frequencies. It is the set of center frequencies of all the points marked as being interfered with; Indicates the first The number of intersections between the frequency hopping sequence and the interference frequency distribution map; This indicates the currently selected frequency hopping sequence. When multiple frequency hopping sequences have the minimum number of intersections, the frequency hopping sequence selection unit selects the frequency hopping sequence with the smallest number in the preset frequency hopping spectrum as the current frequency hopping sequence. The frequency hopping sequence selection unit sends the selected current frequency hopping sequence to the data packet assembly unit and the radio frequency synthesizer control port of the wireless transmission module.

[0038] The data packet assembly unit receives the direct sound field data stream and the scattered sound field data stream, and performs data segmentation and encapsulation according to the frequency hopping timing synchronized with the frequency hopping sequence selection unit. When encapsulating the direct sound field data stream, the data packet assembly unit segments the direct sound field data stream according to a preset first payload length threshold, and loads each segmented data into the payload field of a first data frame. The first payload length threshold is set to 1024 bytes. The data packet assembly unit inserts a first synchronization timestamp into the frame header field of each first data frame. The first synchronization timestamp is provided by the synchronization clock generator inside the wireless transmission module, and its value represents the system clock count value corresponding to the first sampling point in the payload field of that first data frame. All first data frames are arranged in the order of their generation to form a first data packet sequence.

[0039] When encapsulating the scattered sound field data stream, the data packet unit segments the data stream according to a preset second payload length threshold, and loads each segment into the payload field of a second data frame. The second payload length threshold is set to 256 bytes, ensuring that the payload length of the first data frame is greater than that of the second data frame. The data packet unit inserts a second synchronization timestamp into the frame header field of each second data frame. The second synchronization timestamp is provided by the same synchronization clock generator, and its value represents the system clock count value corresponding to the first sampling point in the payload field of that second data frame. All second data frames are arranged in the order of their generation to form a second data packet sequence. During the transmission phase, the radio frequency front-end circuit of the wireless transmission module, based on the first and second data packet sequences output by the data packet unit, alternately or by priority schedules the transmission of the first and second data frames according to the carrier frequency switching order specified by the current frequency hopping sequence.

[0040] Example 3:

[0041] In specific implementation, please refer to Figure 4 The wireless receiving module inside the satellite speaker includes a demodulation and synchronization unit, an attitude calculation unit, a channel mapping extraction unit, and an environment filling generation unit.

[0042] The demodulation and synchronization unit receives radio frequency (RF) signals from the satellite speaker's antenna. The unit incorporates a frequency-hopping sequence generator matched to the main speaker's wireless transmission module. This generator switches the local oscillator frequency according to a frequency-hopping timing synchronized with the main speaker's wireless transmission module. The demodulation and synchronization unit recovers the baseband data frames from the RF signals through mixing and filtering. It then performs frame synchronization detection on the recovered baseband data frames, parsing the synchronization timestamp and data stream type identifier from the frame header field. Based on the data stream type identifier, the unit distributes the baseband data frames to a first data buffer and a second data buffer. The first data buffer stores a first data packet sequence carrying a direct sound field data stream type identifier, and the second data buffer stores a second data packet sequence carrying a scattered sound field data stream type identifier. The demodulation and synchronization unit sorts and deduplicates the first data frames in the first data buffer according to the order of the first synchronization timestamps, and splices them together to form a continuous first synchronization data stream; it sorts and deduplicates the second data frames in the second data buffer according to the order of the second synchronization timestamps, and splices them together to form a continuous second synchronization data stream.

[0043] The attitude calculation unit is connected to the inertial measurement unit (IMU) built into the satellite speaker enclosure via a serial bus. The IMU includes a three-axis accelerometer and a three-axis gyroscope. The attitude calculation unit reads the three-axis accelerometer and gyroscope data output from the IMU at a preset attitude sampling frequency, which can be set to 200Hz. The attitude calculation unit uses a complementary filtering algorithm to fuse the three-axis accelerometer and gyroscope data to calculate the IMU's own attitude quaternion. Based on the attitude quaternion, the attitude calculation unit calculates the IMU's Euler angles, which include pitch, roll, and yaw angles. This set of Euler angles serves as the satellite speaker enclosure's attitude angle data. The attitude calculation unit then provides this attitude angle data to the channel mapping extraction unit.

[0044] The channel mapping extraction unit calculates the real-time azimuth information of the satellite speaker enclosure relative to the main speaker enclosure based on attitude angle data. The unit pre-stores reference azimuth data of the main speaker enclosure in the reference coordinate system, which includes the azimuth angle of the main speaker enclosure relative to the calibration zero point of the inertial measurement unit. The unit algebraically sums the yaw angle from the attitude angle data with the azimuth angle from the reference azimuth data to obtain the azimuth angle value of the satellite speaker enclosure relative to the main speaker enclosure. After projecting corrections to the pitch and roll angles, it finally obtains real-time azimuth information containing both azimuth and elevation components.

[0045] After obtaining real-time azimuth information, the channel mapping extraction unit reads a preset channel mapping table. The preset channel mapping table is a two-dimensional lookup table. The row index corresponds to the quantized value of the azimuth component, and the column index corresponds to the quantized value of the elevation component. Each entry records a target channel identifier. The target channel identifier is a channel number defined in the multi-channel audio coding standard, including left front channel identifier, right front channel identifier, center channel identifier, left surround channel identifier, and right surround channel identifier. The channel mapping extraction unit maps the azimuth and elevation components in the real-time azimuth information to the nearest row index quantized value and column index quantized value, respectively, and reads the corresponding target channel identifier from the preset channel mapping table. The channel mapping extraction unit uses the target channel identifier to extract data from the first synchronous data stream. Each audio data packet in the first synchronous data stream carries the channel identifier of its own channel in the packet header field. The channel mapping extraction unit compares the channel identifier of the audio data packet with the target channel identifier one by one, filters out the audio data packets whose channel identifier matches the target channel identifier, and reads the content of the payload field from the filtered audio data packets as the output channel data.

[0046] The environment padding generation unit receives the second synchronization data stream and extracts environment padding data from it. In one processing method, the environment padding generation unit performs mean filtering on all data packets in the second synchronization data stream. The mean filtering process is as follows: the environment padding generation unit calculates the arithmetic mean of the sampled values ​​of different data packets in the second synchronization data stream at the same time position point by point, and uses the arithmetic mean sequence as the environment padding data.

[0047] In another processing method, the environment filling generation unit performs energy-weighted averaging on all data packets in the second synchronization data stream. The specific implementation of energy-weighted averaging is expressed as follows:

[0048]

[0049] in, Indicates the second synchronous data stream The data packet in the first The sampled values ​​at each sampling time. This is the sequence number of the data packet in the second synchronous data stream. The range of values ​​is to integers, This represents the total number of data packets in the second synchronous data stream. This is the index number of the sampling time. The range of values ​​is from The integer representing the total number of data packet sampling points in the second synchronous data stream; Indicates assignment to the first The weighting coefficient of each data packet; Indicates the first The environmental filling data values ​​are obtained after energy-weighted averaging at each sampling time point. Weighting coefficients. The spatial spread coefficient is determined by the spatial spread factor carried in the data packet. The spatial spread coefficient is a parameter embedded in the header field of each second data frame, and its value ranges from [value missing]. to The environment filling generation unit reads the spatial diffusion coefficient of each second data frame and directly uses the spatial diffusion coefficient as the weight coefficient of the corresponding data packet, i.e. The value is taken as the spatial diffusion coefficient carried by the data packet. The ambient filling generation unit outputs the final channel data and ambient filling data to the power amplifier module simultaneously.

[0050] Example 4:

[0051] In specific implementation, please refer to Figure 5 The power amplifier module acquires channel data and ambient filling data from the wireless receiver module and performs signal merging, gain adjustment and power drive output.

[0052] The power amplifier module has a built-in gain adjustment unit that stores a preset first gain coefficient and a preset second gain coefficient. The first gain coefficient is used to amplify the channel data, and its value is set to 1.0, indicating that the signal amplitude of the channel data is not attenuated. The second gain coefficient is used to amplify the ambient fill data, and its value is set to 0.5, indicating that the signal amplitude of the ambient fill data is attenuated by half, making the second gain coefficient less than the first gain coefficient. In one embodiment, the value of the second gain coefficient ranges from 0.2 to 0.8, and the specific value is set by the system during initial calibration based on the diameter of the full-range speaker unit of the satellite speaker enclosure. When the diameter of the full-range speaker unit is less than 3 inches, the second gain coefficient is 0.6; when the diameter of the full-range speaker unit is greater than or equal to 3 inches, the second gain coefficient is 0.4.

[0053] The gain adjustment unit multiplies each sampled value of the channel data by a first gain coefficient to generate a first amplified signal. The gain adjustment unit multiplies each sampled value of the ambient filling data by a second gain coefficient to generate a second amplified signal. The first and second amplified signals are time-domain aligned.

[0054] The power amplifier module also includes a signal superposition unit. This unit receives a first amplified signal and a second amplified signal, and performs a time-domain superposition operation on them. The time-domain superposition operation is as follows: the signal superposition unit algebraically adds the sampled values ​​of the first amplified signal and the second amplified signal point by point, and the summation results form a sampled sequence of the combined analog signal.

[0055] The signal superposition unit outputs the merged analog signal to the pulse width modulation (PWM) unit inside the power amplifier module. The PWM unit uses pulse width modulation technology to convert the merged analog signal into a drive signal. The PWM unit contains a triangular wave generator that outputs a fixed-frequency carrier triangular wave signal, set to 400kHz. The PWM unit compares the voltage value of the merged analog signal with the instantaneous voltage value of the carrier triangular wave signal in real time. When the voltage value of the merged analog signal is greater than the instantaneous voltage value of the carrier triangular wave signal, the PWM unit outputs a high level; when the voltage value of the merged analog signal is less than or equal to the instantaneous voltage value of the carrier triangular wave signal, the PWM unit outputs a low level. This generates a pulse sequence with a duty cycle varying with the amplitude of the merged analog signal; this pulse sequence is the drive signal. The PWM unit outputs the drive signal to the voice coil of the full-range speaker unit, and the movement of the voice coil in the magnetic field drives the diaphragm of the full-range speaker unit to produce sound.

[0056] In terms of system layout, this system includes a mobile control terminal, which is used to generate satellite speaker layout guidance information. The mobile control terminal is a portable computing device running a layout guidance application, and the portable computing device has a display screen, a touch interface, and a wireless local area network communication module.

[0057] In practice, the mobile control terminal sends a map data request to the wireless transmission module of the main speaker via a wireless local area network communication module, and obtains two-dimensional planar map data of the target listening area from the main speaker or a cloud server. The two-dimensional planar map data is a vector planar map containing the outline of the target listening area, the location of obstacles, and the boundary dimensions.

[0058] After acquiring 2D map data, the layout guidance application on the mobile control terminal displays the 2D map data on the screen and receives user drag-and-drop operations through a touch interface to position the main speaker icon at a specific location on the 2D map data. The layout guidance application reads the pixel coordinates of the main speaker icon on the 2D map data, and obtains the main position coordinates of the main speaker through coordinate transformation. The main position coordinates include the horizontal and vertical coordinates in the world coordinate system defined by the 2D map data.

[0059] The layout guide application reads the preset multi-dimensional surround sound field coverage radius. The preset multi-dimensional surround sound field coverage radius is a length value pre-stored in the mobile control terminal's memory, representing the standard distance between the ideal listening position and the main speaker. The preset multi-dimensional surround sound field coverage radius can be set to 2.5 meters.

[0060] The layout guidance application on the mobile control terminal generates an initial circular layout line on a two-dimensional planar map data, using the main location coordinates as the center and a preset multi-dimensional surround sound field coverage radius as the reference radius. The initial circular layout line is a closed loop curve composed of dense discrete points, and the Euclidean distance from each discrete point to the main location coordinates is equal to the preset multi-dimensional surround sound field coverage radius.

[0061] The layout guidance application on the mobile control terminal extracts obstacle outlines from two-dimensional planar map data. These outlines are sets of polygonal boundary segments representing walls, furniture, or other impenetrable objects. The application performs collision detection on each discrete point on the initial circular layout line. The collision detection process is as follows: the application determines whether a discrete point on the initial circular layout line falls within a closed polygonal region defined by the obstacle outlines. Discrete points falling within closed polygonal regions are marked as unusable, while those not falling within any closed polygonal region are retained. All retained discrete points are clustered into several continuous arc segments based on their continuity on the initial circular layout line. The set of these continuous arc segments constitutes the set of layable arc segments.

[0062] The layout guide application obtains the total number of satellite speakers recorded in the system configuration. The total number of satellite speakers can be obtained from the main speaker via wireless communication or manually entered by the user in the application interface.

[0063] The layout guidance application of the mobile control terminal generates a sequence of candidate satellite speaker positions on the set of layable arc segments using the following method: The layout guidance application calculates the total arc length of all layable arc segments in the set, divides the total arc length evenly into parts equal to the total number of satellite speaker units, and obtains an initial candidate point at each evenly divided point. The generation of the initial candidate point is expressed as follows:

[0064]

[0065] in, This represents the total arc length of the set of layable arc segments, in meters. This indicates the total number of satellite speaker units. It is a positive integer greater than or equal to 2; Indicates the index of the candidate point. The range of values ​​is to Integers; Indicates the first The arc length coordinates of the initial candidate points on the set of layable arc segments; Represents the coordinates of the arc length position on the set of layable arc segments. The function corresponding to the coordinate point at that location outputs the two-dimensional coordinates in the two-dimensional planar map data. Indicates the first The coordinates of the initial candidate points.

[0066] After generating all initial candidate points, the layout guidance application of the mobile control terminal applies a random jitter offset to each of the multiple initial candidate points. The random jitter offset is generated by a pseudo-random number generator invoked by the layout guidance application. The pseudo-random number generator outputs a random vector following a two-dimensional Gaussian distribution with a mean of 0 and a standard deviation of 0.1 meters. The coordinates of each initial candidate point are added to the corresponding random jitter offset to obtain one candidate satellite speaker position from the candidate satellite speaker position sequence.

[0067] The layout guidance application on the mobile control terminal sends the candidate satellite speaker location sequence to the wireless transmission module of the main speaker via the wireless LAN communication module. After receiving the candidate satellite speaker location sequence, and upon confirmation of the layout by the user via the confirmation button on the main speaker or through the mobile control terminal interface, the wireless transmission module of the main speaker generates a location confirmation signal and returns it to the mobile control terminal. Upon receiving the location confirmation signal, the mobile control terminal marks the candidate satellite speaker location sequence on a two-dimensional map, generates and displays satellite speaker layout guidance information on the screen.

[0068] Example 5:

[0069] In specific implementation, please refer to Figure 6 The power management module inside the main speaker enclosure includes a battery power prediction submodule, which is connected to the battery pack in the main speaker enclosure and the RF front-end circuitry of the wireless transmission module. The battery power prediction submodule is used to monitor the battery status in real time and adjust the transmission power of the wireless transmission module.

[0070] The battery power prediction submodule acquires the battery terminal voltage and current values ​​of the main speaker enclosure in real time via an analog front-end interface. The battery terminal voltage is the instantaneous voltage difference between the positive and negative terminals of the main speaker enclosure's battery pack, measured in volts. The battery terminal current is the instantaneous current intensity flowing from the battery pack, measured in amperes. The battery power prediction submodule performs anti-aliasing filtering on the acquired battery terminal voltage and current values. This anti-aliasing filtering is implemented using a second-order Butterworth low-pass filter with a cutoff frequency of 10Hz to remove instantaneous glitches caused by sudden load changes.

[0071] The battery power prediction submodule calculates the current remaining battery percentage based on the filtered battery terminal voltage and current values. The calculation employs an estimation algorithm combining coulomb counting and open-circuit voltage correction. The coulomb counting part integrates the battery terminal current over time to obtain the amount of charge consumed since full charge; the difference between the consumed charge and the nominal total charge of the battery pack is the remaining charge. The open-circuit voltage correction part searches for the corresponding corrected state of charge value on the battery pack's preset voltage-state-of-charge curve based on the battery terminal voltage value. The battery power prediction submodule then performs a weighted fusion of the remaining charge and the corrected state of charge value, with weighting coefficients of 0.7 and 0.3 respectively. The fusion result is converted into a percentage value relative to the nominal total charge of the battery pack, which is the current remaining battery percentage.

[0072] The battery power prediction submodule internally stores a preset power level lookup table. This table is a one-dimensional mapping table; its input is the range of the current remaining battery percentage, and its output is the transmission power level. The specific mapping relationships are as follows: when the current remaining battery percentage is between 100% and 60%, the corresponding transmission power level is level 0, with a transmission power of +20dBm for the RF front-end circuit; when the current remaining battery percentage is between 59% and 30%, the corresponding transmission power level is level 1, with a transmission power of +15dBm for the RF front-end circuit; when the current remaining battery percentage is between 29% and 10%, the corresponding transmission power level is level 2, with a transmission power of +10dBm for the RF front-end circuit; and when the current remaining battery percentage is between 9% and 0%, the corresponding transmission power level is level 3, with a transmission power of +5dBm for the RF front-end circuit. The battery power prediction submodule performs a range matching query in a preset power level lookup table based on the calculated current remaining power percentage, and returns the transmission power level corresponding to the range in which the current remaining power percentage is located.

[0073] The battery power prediction submodule generates a power control command, which is a 4-bit digital control word containing a transmit power level value. The battery power prediction submodule sends the power control command to the radio frequency (RF) front-end circuit of the wireless transmitter module via an integrated circuit bus interface. The RF front-end circuit internally contains a programmable gain power amplifier. Based on the received power control command, the programmable gain power amplifier adjusts the bias voltage and gain level to set the output power of the transmitted signal to the power value corresponding to the transmit power level.

[0074] Regarding sound field energy distribution, the audio processing module includes a rhythm analysis unit, which dynamically adjusts the energy distribution ratio between the direct sound field data stream and the scattered sound field data stream. The rhythm analysis unit receives the raw multi-channel audio data and performs rhythmic feature analysis on it. The rhythm feature analysis process is as follows: The rhythm analysis unit downmixes the original multi-channel audio data, merging it into a mono audio signal; the rhythm analysis unit performs a short-time Fourier transform on the mono audio signal, using a Hanning window with a window length of 2048 sampling points and a frame shift step of 512 sampling points; the rhythm analysis unit extracts the low-frequency energy envelope of each spectral frame from the amplitude spectrum of the short-time Fourier transform, with the frequency range of the low-frequency energy envelope being 20Hz to 200Hz; the rhythm analysis unit performs autocorrelation calculation on the low-frequency energy envelope, and the reciprocal of the time interval corresponding to the peak value of the autocorrelation function obtained from the autocorrelation calculation is the number of beats per minute, and the rhythm analysis unit uses the number of beats per minute as the beat density value of the current audio segment.

[0075] The rhythm analysis unit compares the beat density value with preset high-density thresholds and preset low-density thresholds. The preset high-density threshold is set to 120 beats per minute, and the preset low-density threshold is set to 60 beats per minute. When the beat density value is greater than the preset high-density threshold, it indicates that the rhythm of the current audio segment is relatively intense, and the audio processing module adjusts the energy allocation ratio to the first allocation ratio. In the first allocation ratio, the energy proportion of the direct sound field data stream is set to 0.75, and the energy proportion of the scattered sound field data stream is set to 0.25, so that the energy proportion of the direct sound field data stream is greater than that of the scattered sound field data stream.

[0076] When the beat density value is less than the preset low density threshold, it indicates that the rhythm of the current audio segment is relatively slow, and the audio processing module adjusts the energy allocation ratio to the second allocation ratio. In the second allocation ratio, the energy proportion of the direct sound field data stream is set to 0.35, and the energy proportion of the scattered sound field data stream is set to 0.65, so that the energy proportion of the direct sound field data stream is less than the energy proportion of the scattered sound field data stream.

[0077] When the beat density value is between a preset low-density threshold and a preset high-density threshold, i.e., greater than or equal to 60 beats per minute and less than or equal to 120 beats per minute, the audio processing module maintains the energy allocation ratio at a preset default allocation ratio. In this default allocation ratio, the energy proportion of the direct sound field data stream is set to 0.55, and the energy proportion of the scattered sound field data stream is set to 0.45. Based on the adjusted energy allocation ratio, the audio processing module applies corresponding energy scaling factors to the spectral coefficients of the direct and scattered sound field data streams in the frequency domain, thereby achieving dynamic control of the energy of the two data streams.

[0078] For dynamic sound pressure level compensation, the satellite speaker enclosure includes a distance sensing submodule, which is connected to the power amplifier module via a serial data interface. The distance sensing submodule contains an ultrasonic transducer and a timing measurement circuit. It emits an ultrasonic ranging signal, a pulse train with a center frequency of 40kHz and a pulse width of 1 millisecond. The ultrasonic transducer converts the electrical pulse signal into ultrasonic waves that radiate towards the listening position, while the timing measurement circuit records the transmission time of the ultrasonic ranging signal. When an ultrasonic ranging signal encounters an obstacle in its propagation path, it is reflected. The reflected echo is received by the same ultrasonic transducer and converted into an electrical signal. The timing and measurement circuit records the moment the ultrasonic echo signal is received. .

[0079] The distance sensing submodule calculates the real-time distance between the satellite speaker and the listening position. The formula for calculating the real-time distance is as follows:

[0080]

[0081] in, Indicates the transmission time of the ultrasonic ranging signal, in seconds; Indicates the moment the ultrasonic echo signal was received, in seconds; This represents the speed of sound in air at the current ambient temperature, measured in meters per second. The value is obtained through the formula calculate, The ambient temperature value is measured by the temperature sensor built into the distance sensing submodule, in degrees Celsius. This represents the calculated real-time distance between the satellite speaker enclosure and the listening position, in meters.

[0082] The distance sensing submodule sends the calculated real-time distance value to the power amplifier module. The power amplifier module internally stores a preset distance-sound pressure compensation curve. This preset curve is implemented as a lookup table, with the distance value as input and the compensation gain value as output. The mapping rule for the preset distance-sound pressure compensation curve is: for every 1 meter increase in distance, the compensation gain increases by 6 dB. The power amplifier module uses the real-time distance value as an index to look up the distance-sound pressure compensation curve in the lookup table. If the real-time distance value falls between two entries, linear interpolation is used to calculate the corresponding compensation gain value. The power amplifier module multiplies the amplitude of the current drive signal by the found compensation gain value to obtain the final drive signal. The amplitude multiplication operation is achieved by proportionally adjusting the duty cycle of the drive signal pulse width modulation waveform. The compensation gain value is converted into a multiplicative correction factor for the duty cycle and applied to the pulse width modulation unit. The power amplifier module drives the full-range speaker unit with the final drive signal. The sound waves radiated by the full-range speaker unit, after propagation and attenuation due to the real-time distance, reach the preset target sound pressure level at the listening position.

[0083] The above description is merely a specific embodiment of this application, but the scope of protection of this application is not limited thereto. Any changes or substitutions that can be easily conceived by those skilled in the art within the scope of the technology disclosed in this application should be included within the scope of protection of this application.

Claims

1. A portable multi-dimensional surround audio playback system, characterized in that, The system includes: The main speaker enclosure integrates an audio processing module, a wireless transmission module, and a power management module. Multiple satellite speaker enclosures, each integrating a wireless receiver module, a power amplifier module, and a full-range speaker unit; The audio processing module is used to receive raw multi-channel audio data and decompose the raw multi-channel audio data into direct sound field data stream and scattered sound field data stream based on the sound field spatial feature model. The wireless transmission module is used to package the direct sound field data stream into a first data packet sequence and the scattered sound field data stream into a second data packet sequence; The wireless receiving module is used to receive the first data packet sequence and the second data packet sequence, and extract channel data corresponding to the real-time orientation information from the first data packet sequence according to the real-time orientation information of the satellite speaker relative to the main speaker, and extract environmental filling data from the second data packet sequence. The power amplification module is used to combine the channel data and the ambient filling data into a driving signal to drive the full-range speaker unit to produce sound.

2. The portable multi-dimensional surround audio playback system according to claim 1, characterized in that, The audio processing module receives raw multi-channel audio data and, based on a sound field spatial feature model, decomposes the raw multi-channel audio data into a direct sound field data stream and a scattered sound field data stream, specifically including: The original multi-channel audio data is subjected to frame segmentation processing to obtain multiple audio frame data blocks; Perform a time-frequency transformation on each audio frame data block to obtain the frequency domain representation matrix corresponding to each audio frame data block; Based on the predefined direction vector dictionary in the sound field spatial feature model, the frequency domain representation matrix is ​​spatially sparsely decomposed to obtain directional sound field components and environmental sound field components. The temporal reconstruction signal of the directional sound field component is labeled as the direct sound field data stream, and the temporal reconstruction signal of the ambient sound field component is labeled as the scattered sound field data stream.

3. A portable multi-dimensional surround audio playback system according to claim 2, characterized in that, When the audio processing module performs time-frequency transformation on each audio frame data block, it uses modified discrete cosine transform to map each audio frame data block from the time domain to the frequency domain, and outputs the frequency domain coefficient matrix as the frequency domain representation matrix.

4. A portable multi-dimensional surround audio playback system according to claim 1, characterized in that, The wireless transmission module packages the direct sound field data stream into a first data packet sequence and the scattered sound field data stream into a second data packet sequence, specifically including: Obtain the interference frequency distribution map in the current network environment; Based on the interference frequency distribution map, the current frequency hopping sequence is selected from the preset frequency hopping spectrum; Using the current frequency hopping sequence, the direct sound field data stream is segmented and encapsulated into multiple first data frames, and a first synchronization timestamp is inserted into the frame header of each first data frame to obtain the first data packet sequence; Using the current frequency hopping sequence, the scattered sound field data stream is segmented and encapsulated into multiple second data frames, and a second synchronization timestamp is inserted into the frame header of each second data frame to obtain the second data packet sequence; The payload length of the first data frame is greater than the payload length of the second data frame.

5. A portable multi-dimensional surround audio playback system according to claim 4, characterized in that, When selecting the current frequency hopping sequence based on the interference frequency distribution map, the wireless transmission module preferentially selects the frequency hopping sequence with the fewest intersections with interference frequency points in the interference frequency distribution map as the current frequency hopping sequence.

6. A portable multi-dimensional surround audio playback system according to claim 1, characterized in that, The wireless receiving module receives the first data packet sequence and the second data packet sequence, and extracts channel data corresponding to the real-time location information from the first data packet sequence based on the real-time location information of the satellite speaker relative to the main speaker, and simultaneously extracts environmental filling data from the second data packet sequence, specifically including: The received first data packet sequence and second data packet sequence are synchronously demodulated to obtain a first synchronous data stream and a second synchronous data stream; The attitude angle data output by the inertial measurement unit built into the satellite speaker enclosure is read, and the real-time orientation information of the satellite speaker enclosure relative to the main speaker enclosure is calculated based on the attitude angle data. Based on the real-time location information, the target channel identifier is queried from the preset channel mapping table; Based on the target channel identifier, audio data packets carrying the target channel identifier are filtered out from the first synchronous data stream, and the payload within the audio data packets is extracted as the channel data; The environmental filling data is obtained by performing mean filtering on all data packets in the second synchronization data stream.

7. A portable multi-dimensional surround audio playback system according to claim 6, characterized in that, The wireless receiving module performs energy-weighted averaging on all data packets in the second synchronization data stream and assigns weight coefficients according to the spatial diffusion coefficient carried by each data packet to obtain the environmental filling data.

8. A portable multi-dimensional surround audio playback system according to claim 1, characterized in that, The power amplification module combines the channel data and the ambient filling data into a driving signal to drive the full-range speaker unit to produce sound, specifically including: A first gain coefficient is applied to the audio channel data to obtain a first amplified signal; A second gain coefficient is applied to the environmental filling data to obtain a second amplified signal, wherein the second gain coefficient is smaller than the first gain coefficient; The first amplified signal and the second amplified signal are time-domain superimposed to obtain a merged analog signal; The combined analog signal is converted into the driving signal using pulse width modulation technology, and the driving signal is output to the voice coil of the full-range loudspeaker unit.

9. A portable multi-dimensional surround audio playback system according to claim 1, characterized in that, The system also includes a mobile control terminal, which is used to generate satellite speaker layout guidance information, specifically including: The mobile control terminal acquires two-dimensional planar map data of the target listening area; The mobile control terminal marks the main position coordinates of the main speaker on the two-dimensional planar map data; The mobile control terminal generates a sequence of candidate satellite speaker locations on the two-dimensional plane map data based on the main location coordinates and the preset multi-dimensional surround sound field coverage radius. The mobile control terminal sends the candidate satellite speaker location sequence to the wireless transmission module of the main speaker body, and receives the location confirmation signal returned by the wireless transmission module to generate the satellite speaker layout guidance information.

10. A portable multi-dimensional surround audio playback system according to claim 9, characterized in that, The mobile control terminal generates a sequence of candidate satellite speaker locations on the two-dimensional plane map data based on the main location coordinates and a preset multi-dimensional surround sound field coverage radius, specifically including: The mobile control terminal generates an initial circular layout line on the two-dimensional plane map data with the main position coordinates as the center and the preset multi-dimensional surround sound field coverage radius as the reference radius. The mobile control terminal performs collision detection on the initial circular layout line based on the obstacle outlines in the two-dimensional planar map data to obtain a set of layable arc segments; The mobile control terminal samples at equal angular intervals on the set of layable arc segments based on the total number of satellite speaker units to obtain multiple initial candidate points. The mobile control terminal applies a random jitter offset to the plurality of initial candidate points to generate the candidate satellite speaker position sequence.

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