A method for seamlessly bridging traditional audio devices with Auracast

Abstract

This invention discloses a method for seamlessly bridging conventional audio devices with Auracast, relating to the fields of audio signal processing and wireless communication technology. By identifying the personalized audio parameters of conventional audio devices and generating dedicated configuration files, it achieves deep customization of the audio signal conversion process. This method goes beyond simple format conversion; it performs targeted signal adaptation and correction based on the physical characteristics of each device, such as input impedance and frequency response. This maximizes the sound quality potential of conventional devices and avoids distortion and frequency response defects caused by parameter mismatches in general solutions, significantly improving the fidelity of audio output and the listening experience.

Description

Technical Field

[0001] This invention relates to the fields of audio signal processing and wireless communication technology, and specifically to a method for seamlessly bridging conventional audio devices with Auracast. Background Technology

[0002] Auracast is a next-generation audio broadcasting feature based on Bluetooth Low Energy audio technology. It allows an audio source device, such as a smartphone or television, to broadcast to an unlimited number of Bluetooth audio receiving devices, such as headphones or speakers, within its range, simultaneously broadcasting one or more audio streams. This one-to-many broadcasting mode provides innovative solutions for scenarios such as audio sharing in public places, hearing assistance, and multilingual simultaneous interpretation. However, there are still many traditional audio devices on the market that lack Bluetooth receiving capabilities, especially Auracast functionality, such as wired headphones and older sound systems. These devices cannot directly receive and play Auracast audio broadcasts.

[0003] Currently, the common technical solution for enabling traditional audio devices to receive wireless audio signals is to use a universal Bluetooth audio receiver. This receiver receives standard Bluetooth audio streams, converts them into analog audio signals using internal digital-to-analog converter circuitry, and then outputs them to traditional audio devices through a standard audio interface. This approach solves the connectivity issue to some extent, but its design is typically standardized, aiming to be compatible with as many devices as possible, rather than being optimized for specific devices.

[0004] Existing technical solutions have significant technical shortcomings in practical applications. First, general-purpose Bluetooth receivers typically have high audio latency, and when multiple receivers are connected, the lack of an effective synchronization mechanism leads to echoes when multiple traditional devices play simultaneously. Second, the output signal characteristics of these receivers are fixed and cannot adaptively adjust to the electrical characteristics of the connected traditional audio devices. This often results in volume mismatch, frequency response distortion, or dynamic range compression, failing to fully utilize the sound quality performance of traditional devices. Finally, the connection and pairing process may still be relatively complex for non-technical users, failing to achieve a seamless and convenient access experience. Summary of the Invention

[0005] The purpose of this invention is to provide a method for seamlessly bridging conventional audio devices with Auracast, thereby solving the problems existing in the prior art.

[0006] To solve the above-mentioned technical problems, the present invention adopts the following technical solution: The present invention provides a method for seamlessly bridging traditional audio devices with Auracast, comprising: S1. scanning Auracast broadcast sources and traditional audio devices in the surrounding environment to generate a device list, wherein the device list includes the identifier, input impedance parameters, supported frequency band parameters, maximum bit depth, maximum supported sampling rate, device delay characteristic parameters, frequency response data, dedicated static audio parameters, hardware tolerance distortion threshold, multi-band equalizer gain parameters, and output gain parameters of each traditional audio device.

[0007] S2. Based on the device list, identify the audio setting parameters of each traditional audio device and generate a configuration file for each traditional audio device.

[0008] S3. Based on the configuration files of each traditional audio device, convert the Auracast audio data stream into compatible audio data for each traditional audio device.

[0009] S4. Perform dynamic buffer adjustment on the dedicated filtered audio data after dedicated digital equalization filtering and dynamic range compression correction for the compatible audio data of each traditional audio device, and generate buffered audio data for each traditional audio device.

[0010] S5. Apply synchronization control to the buffered audio data of each traditional audio device to generate synchronized audio output from all traditional audio devices.

[0011] The beneficial effects of this invention are as follows: By identifying the personalized audio parameters of traditional audio devices and generating exclusive configuration files, this invention achieves deep customization of the audio signal conversion process. This method goes beyond simple format conversion; it performs targeted signal adaptation and correction based on the physical characteristics of each device, such as input impedance and frequency response. This maximizes the sound quality potential of traditional devices, avoiding distortion and frequency response defects caused by parameter mismatches in general solutions, and significantly improving the fidelity of audio output and the listening experience.

[0012] This invention integrates dynamic buffer adjustment and multi-device synchronization control technologies, effectively solving the inherent latency and timing inconsistencies in audio transmission and processing links in wireless broadcasting. By dynamically adjusting the data buffer based on real-time network conditions and device processing latency, continuous and smooth playback of the audio stream is ensured. Simultaneously, through a precise synchronization control mechanism, the playback clocks of multiple terminal devices are aligned, eliminating echoes and phase interference during multi-device playback, achieving a highly consistent synchronized sound field, and guaranteeing playback stability and synchronization in complex application scenarios.

[0013] This invention significantly simplifies the user's operation process and provides clear and intuitive status feedback, greatly improving product usability and user experience. By automating a series of complex technical steps such as device scanning, identification, configuration, and connection, and encapsulating them into a single user command, users can complete the bridging process without specialized knowledge. Simultaneously, visual and audible status feedback makes the entire process transparent, enhancing the user's sense of control and trust in the system, enabling advanced audio sharing technology to conveniently benefit users with traditional devices. Attached Figure Description

[0014] To more clearly illustrate the technical solutions in the embodiments of the present invention or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present invention. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0015] Figure 1 This is a schematic diagram of the implementation steps of the method of the present invention.

[0016] Figure 2 This is a flowchart illustrating the scanning and identification process for bridging devices. Detailed Implementation

[0017] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0018] Reference Figure 1 As shown, the present invention provides a method for seamlessly bridging conventional audio devices with Auracast, comprising: S1. scanning Auracast broadcast sources and conventional audio devices in the surrounding environment to generate a device list, wherein the device list includes the identifier, input impedance parameters, supported frequency band parameters, maximum bit depth, maximum supported sampling rate, device delay characteristic parameters, frequency response data, proprietary static audio parameters, hardware tolerance distortion threshold, multi-band equalizer gain parameters, and output gain parameters of each conventional audio device.

[0019] Reference Figure 2As shown, it should be noted that the traditional audio device and its corresponding identifier, input impedance parameters, and supported frequency band parameters are obtained through an audio characteristic detection process. A preset wideband test signal is sent to the traditional audio device through the audio output interface of the bridging device, and the voltage and current changes at the audio output interface are measured simultaneously. The input impedance parameters are calculated by combining the frequency and voltage / current response relationship of the test signal. Simultaneously, the response amplitude of the traditional audio device to the wideband test signal is analyzed to determine the gain or attenuation at different frequencies, thereby depicting the frequency response data of the traditional audio device. Based on a preset amplitude attenuation threshold, the supported frequency band parameters are extracted from the frequency response data. At the same time, the identifier, maximum bit depth, maximum supported sampling rate, device delay characteristic parameters, proprietary static audio parameters, and hardware distortion tolerance threshold are read.

[0020] S2. Based on the device list, identify the audio setting parameters of each traditional audio device and generate a configuration file for each traditional audio device.

[0021] In a specific embodiment of the present invention, step S2 includes: S2.1. Extracting the identifier, input impedance parameters and supported frequency band parameters of each conventional audio device from the device list.

[0022] S2.2. Import the input impedance parameters and supported frequency band parameters of each traditional audio device into the audio setting parameter model, and output the matching audio setting parameters for each traditional audio device. The audio setting parameter model is as follows: .

[0023] in, For the generated first Matching audio settings parameters for a traditional audio device For the first The input impedance parameters of a traditional audio device For the first The frequency band parameters supported by a traditional audio device The preset mapping rule set is used, and the audio setting parameters are a set of adjustment commands, where r is the number of the traditional audio device. , The number of traditional audio devices is specified, and the audio settings parameters include output gain parameters and multi-band equalizer gain parameters.

[0024] It should be noted that the preset mapping rule set is specifically constructed based on the hardware characteristic database and acoustic performance test data of similar traditional audio devices, and is pre-stored in the data warehouse of the bridging device. It contains multiple sets of hardware parameter range-audio setting parameter association mapping relationships. For example, when >10kΩ, output gain increases by 3dB; when When the low-frequency cutoff frequency is >100Hz, the equalization gain in the 60Hz to 200Hz band is increased by 4dB.

[0025] S2.3. Extract the highest bit depth, maximum supported sampling rate, device delay characteristic parameters, frequency response data, dedicated static audio parameters and hardware tolerance distortion threshold of each traditional audio device from the device list. Combine the matching audio setting parameters, input impedance parameters and supported frequency band parameters of each traditional audio device to generate a configuration file for each traditional audio device. The configuration file is associated with and stored with the identifier of the corresponding traditional audio device.

[0026] By performing the above steps, this method achieves a technological leap from passive adaptation to proactive optimization. It goes beyond simply identifying traditional audio devices; instead, through precise measurement and intelligent calculation, it tailors a unique audio compensation scheme for each device. During subsequent audio conversion and transmission, it can proactively eliminate potential sound quality degradation caused by incompatibility in device physical characteristics, such as insufficient volume, harsh high frequencies, or missing low frequencies. The configuration file, as the core data foundation, ensures that all processing stages in the bridging link work collaboratively, resulting in a final output sound quality far exceeding the simple combinations achievable by independent, non-cooperative adaptation technologies. This achieves a deep exploration of the potential of traditional audio devices and a significant improvement in sound quality.

[0027] S3. Based on the configuration files of each traditional audio device, convert the Auracast audio data stream into compatible audio data for each traditional audio device.

[0028] It should be noted that the conversion process is based on the configuration files of various traditional audio devices: First, the maximum bit depth, maximum supported sampling rate, and supported frequency band parameters in the configuration file are read to adjust the format of the Auracast audio data stream to the range initially recognized by the traditional audio devices. For example, the original 24bit / 96kHz Auracast data stream is initially converted to the 16bit / 44.1kHz format, which is the highest supported format by traditional audio devices, to avoid exceeding the hardware processing limit. At the same time, the multi-band equalizer gain parameters in the configuration file are called to perform preliminary correction on the frequency band characteristics of the Auracast audio data stream, such as compensating for the attenuation range within the frequency band supported by the device in advance. This ensures that the generated compatible audio data can be received normally by traditional audio devices and provides basic adaptation conditions for subsequent steps S7 and S8, avoiding subsequent processing failures due to initial format incompatibility.

[0029] S4. Perform dynamic buffer adjustment on the dedicated filtered audio data after dedicated digital equalization filtering and dynamic range compression correction for the compatible audio data of each traditional audio device, and generate buffered audio data for each traditional audio device.

[0030] In a specific embodiment of the present invention, step S4 includes: S4.1. For each conventional audio device, obtain network condition parameters of the network transmission link and obtain device latency characteristic parameters from the configuration file of each conventional audio device. The network condition parameters include the maximum network jitter value and packet loss rate monitored in the previous observation period. The device latency characteristic parameters include multiple latency parameters inherent to each conventional audio device when processing audio data.

[0031] It should be noted that the maximum network jitter value refers to the maximum deviation between the actual transmission delay and the average transmission delay of Auracast audio data in the network transmission link. For example, if the average delay is 20ms, and the transmission delay is 25ms and 18ms respectively, then the maximum network jitter value is 7ms. The maximum network jitter value directly affects the timing stability of audio data arriving at traditional audio devices. The larger the maximum network jitter value, the more likely it is to cause playback stuttering. The packet loss rate refers to the percentage of audio data packets that fail to be transmitted to traditional audio devices out of the total number of transmitted data packets. For example, if 100 data packets are transmitted and 3 data packets are lost, then the packet loss rate is 3%. The packet loss rate determines the integrity of audio data. An excessively high packet loss rate will lead to audio disconnection or noise.

[0032] It should be noted that the device delay characteristics parameters include signal conversion delay, hardware response delay, etc.

[0033] S4.2. For each traditional audio device, based on its network condition parameters and device latency characteristics, calculate the dynamic buffer adjustment parameters using a buffer adjustment parameter calculation model. .

[0034] in, For the calculated first The dynamic buffer adjustment parameter of a traditional audio device, namely the target buffer duration. To process the first read from the device configuration file The comprehensive characterization value obtained from the device delay characteristic parameters of a traditional audio device. , These are the maximum network jitter value and packet loss rate detected in the previous observation period, respectively. A dimensionless safety factor greater than 1 is used, specifically determined through a device latency-network jitter and dimensionless safety factor adaptation table, to provide a buffer margin based on the maximum jitter. This is a gain function of the preset packet loss rate versus time.

[0035] It should be noted that the first [item] read from the configuration file of the processing device The comprehensive characterization value of the device delay characteristic parameters of a traditional audio device is obtained by weighted summation of multiple delay parameters in the device delay characteristic parameters. The existing technology for weighted summation is relatively mature, so it will not be described in detail here.

[0036] It should be noted that the device latency-network jitter and dimensionless safety factor adaptation table is pre-set based on hardware latency test data of traditional audio devices and network jitter simulation data of multiple scenarios.

[0037] For example, when ≤30ms, and When >20ms, Take 1.8, when >50ms and When ≤10ms, Take 1.3, which means... The product of the maximum network jitter value and the network fluctuation is compensated, and the delay is limited by the device latency characteristic parameters. Increase the size excessively to ensure that the buffer adjustment balances smoothness and low latency.

[0038] It should also be noted that, The dimensionless packet loss rate is converted into a compensatory time increment. Network packet loss test data based on the Auracast broadcast scenario is pre-stored in the bridging device's data warehouse. For example, the gain function is defined as... Its core function is to convert the dimensionless data packet loss rate into a compensatory time increment, in milliseconds, and to follow the adaptation logic that the higher the data packet loss rate, the larger the time increment, in order to cope with possible data retransmission or difference repair needs.

[0039] S4.3. For the compatible audio data of each traditional audio device, apply the calculated buffer adjustment parameters to configure an independent first-in-first-out data buffer for processing, and obtain the fill amount of the data buffer of each traditional audio device in real time. Based on the fill amount of the data buffer of each traditional audio device and the deviation of the target buffer duration (i.e., the fill amount deviation), and combined with the relationship between the fill amount deviation value and the read / write speed stored in the data warehouse, dynamically adjust the reading speed of the compatible audio data of the corresponding data buffer to generate the buffered audio data of each traditional audio device.

[0040] It should be noted that the relationship between the fill volume deviation value and the read / write speed is specifically based on a pre-built correlation model of the traditional audio device hardware read / write capabilities, audio sampling rate, and buffer characteristics, which is pre-stored in the bridging device's data warehouse. The fill volume deviation value is positively correlated with the read / write speed. When the fill volume deviation value is less than 0, the data reading speed is slowed down or more data is waited for to be written. The larger the deviation value is less than 0, the slower the read / write speed is to prevent playback interruption. When the fill volume deviation value is greater than 0, the reading speed is accelerated to reduce overall latency. The larger the deviation value is greater than 0, the faster the read / write speed is. Through this negative feedback adjustment mechanism, the amount of data in the data buffer is always dynamically maintained near the optimal level. Finally, the data flowing out of the buffer and supplied to subsequent steps is stable and smooth buffered audio data.

[0041] By performing the aforementioned dynamic buffer adjustment steps, this method achieves intelligent management of audio data streams. It combines the uncertainties of external network conditions with the inherent fixed latency characteristics of traditional audio devices, thus avoiding the drawbacks of traditional fixed-size buffers. Excessively large buffers lead to unnecessary latency, while excessively small buffers cannot cope with network fluctuations. The technical effect of this method is that it adaptively balances playback smoothness and latency according to real-time environmental changes. It can ensure audio continuity by increasing buffering under poor network conditions, and provide a near-real-time listening experience by reducing buffering under good network conditions. The synergistic effect of this dynamic balancing strategy makes the overall performance and user experience of audio bridging significantly superior to solutions that only consider network conditions or device latency, achieving a balance between stability and low latency.

[0042] S5. Apply synchronization control to the buffered audio data of each traditional audio device to generate synchronized audio output from all traditional audio devices.

[0043] In a specific embodiment of the present invention, step S5 includes: S5.1. Obtaining the output timestamp parameter of each conventional audio device.

[0044] S5.2. From all traditional audio devices, designate the first successfully connected traditional audio device as the master device, and the remaining traditional audio devices as slave devices. For each slave device, compare its output timestamp parameter with the output timestamp parameter of the master device, and calculate the synchronization offset parameter for each slave device separately. The calculation formula is as follows: .

[0045] in, This is the synchronization offset parameter of the m-th slave device, i.e., the playback rate adjustment factor. The current master-slave clock offset for the m-th slave device is calculated by comparing the timestamps output by the m-th slave device and the master device. This is the dedicated integral value of the master-slave clock offset accumulated over time for the m-th slave device. and These are the proportional gain coefficient and the integral gain coefficient, For traditional audio equipment, , The total number of traditional audio devices, It is a subset of r, and .

[0046] It should be noted that the proportional gain coefficient and integral gain coefficient are used to control the response speed and stability of the adjustment. They are obtained through the set of proportional gain coefficients and integral gain coefficients stored in the database. Based on the clock deviation characteristics of the slave device, such as clock drift rate and hardware crystal oscillator accuracy, they are dynamically adapted to adjust the influence weight of the clock deviation proportional term (instantaneous deviation) and integral term (cumulative deviation) on the playback rate adjustment factor, so as to ensure the master-slave clock synchronization accuracy.

[0047] It should also be noted that the set of proportional gain coefficients and integral gain coefficients is constructed based on a database of clock characteristics of traditional audio devices: Classified by hardware type of traditional audio devices such as headphones and desktop speakers, typical clock drift rates and crystal oscillator accuracy parameters of various traditional audio devices are extracted, and corresponding proportional gain coefficients and integral gain coefficients are pre-stored. For example, for headphones... and Desktop speaker devices (default) and 3. The proportional gain coefficient and integral gain coefficient are stored in the bridging device data warehouse. When called, the corresponding proportional gain coefficient and integral gain coefficient are directly read by matching the type according to the slave device number.

[0048] S5.3. For each slave device, use its own dedicated synchronization offset parameter to individually correct the playback timing of the buffered audio data of each slave device, while maintaining the baseline playback timing of the buffered audio data of the master device, and generating synchronized audio output for all traditional audio devices.

[0049] It should be noted that the specific content of individually correcting the playback timing of the buffered audio data of each slave device is as follows: when the synchronization offset parameter of a slave device is greater than 1, it indicates that the local clock of the slave device is lagging behind the master device, and the audio resampling module of the slave device will be adjusted to speed up its playback speed by a preset first amplitude. When the synchronization offset parameter of a slave device is less than 1, it indicates that the local clock of the slave device is ahead of the master device, and the playback speed of the device will be slowed down by a preset second amplitude. The adjustment process of each device is a continuous and subtle independent operation to ensure that the human ear cannot perceive the pitch change of a single device, while achieving the timing uniformity of all devices.

[0050] For example, the first amplitude corresponds to the scenario where the slave device clock lags behind the master device, and is used to speed up the playback speed. It is preset to 0.3% to 0.5%. For example, when the original playback speed is 44.1kHz, it is adjusted to 44.2323kHz to 44.3205kHz, which is suitable for the mild synchronization requirement in the lag scenario. The second amplitude corresponds to the scenario where the slave device clock leads the master device, and is used to slow down the playback speed. It is preset to 0.2% to 0.4%. For example, when the original playback speed is 44.1kHz, it is adjusted to 44.0118kHz to 43.9236kHz, which avoids excessive adjustment that leads to timing fluctuations. Both are continuous and subtle values, and are independently matched to different slave devices, taking into account both synchronization accuracy and listening experience.

[0051] The technical effect of this method is to achieve high-precision audio synchronization between multiple traditional audio devices distributed in different physical locations. It solves the problem of minute but cumulative clock drift caused by differences in the independent crystal oscillators of each device. Without this synchronization control, when multiple devices play the same audio simultaneously, audible echoes or phase interference effects gradually develop, severely damaging the listening experience. Through the synergistic effect of master-slave clock synchronization and dynamic playback rate adjustment, this method transforms multiple independent traditional speakers into a logically unified and synchronized audio playback system. This complements and enhances the aforementioned dynamic buffering step; dynamic buffering ensures the continuity of the data stream, while synchronization control ensures the temporal consistency of the data stream at the playback endpoint. The combination of the two achieves smooth and synchronized high-quality audio broadcasting.

[0052] In a specific embodiment of the present invention, the method for seamlessly bridging conventional audio devices with Auracast further includes the following step: S6. Determine dedicated optimized transmission parameters for each conventional audio device.

[0053] S7. For each traditional audio device, the application uses its own optimized transmission parameters to perform parameter adaptation processing on the compatible audio data corresponding to each traditional audio device. Specifically, the parameter adaptation processing involves resampling or adjusting the bit depth of the compatible audio data to generate exclusive adapted audio data for each traditional audio device.

[0054] Specifically, if the sampling rate adjustment parameter of a traditional audio device, i.e. the maximum supported sampling rate, is lower than the initial sampling rate of the acquired compatible audio data, then the compatible audio data of the traditional audio device is resampled separately according to the sampling rate adjustment parameter to ensure that the sampling rate of the resampled data does not exceed the maximum supported capacity of the device, and to avoid device overload, distortion or playback interruption due to excessively high sampling rate.

[0055] If the maximum bit depth of a traditional audio device does not match the bit depth of the acquired compatible audio data, then the bit depth adjustment parameter in the dedicated optimized transmission parameters is used to perform bit depth adjustment processing on the compatible audio data of the traditional audio device separately: when the bit depth of the compatible audio data is higher than the maximum bit depth of the traditional audio device, bit depth downsampling processing is performed according to the target bit depth adjustment parameter, such as using a jitter quantization algorithm to reduce the sound quality loss caused by downsampling, and at the same time, the audio data bit rate is reduced synchronously according to the bit rate adjustment parameter.

[0056] When the bit depth of the compatible audio data is lower than the maximum bit depth of the traditional audio device and the dedicated optimized transmission parameters determine that it needs to be increased to the optimal bit depth supported by the traditional audio device, i.e., the target bit depth adjustment parameter, bit depth upsampling is performed according to the target bit depth adjustment parameter, and the audio data bit rate is simultaneously increased according to the bit rate adjustment parameter to fully realize the sound quality reproduction potential of the traditional audio device.

[0057] S8. Perform adaptive filtering processing separately for the dedicated audio data of each traditional audio device: call the frequency response data in the configuration file of each traditional audio device to generate a dedicated digital equalization filter that is exactly opposite to the inherent frequency response curve of each traditional audio device. At the same time, combine the real-time audio quality feedback parameters of each traditional audio device to enable a dedicated dynamic range compressor for each traditional audio device, generate dedicated filtered audio data for each traditional audio device, and use the dedicated filtered audio data of each traditional audio device as the input of step S4 for each traditional audio device to ensure that the buffer adjustment of each traditional audio device is based on its own corrected high-quality audio data.

[0058] By performing the above steps, this method achieves a transformation from simple signal format conversion to deep, personalized sound quality reshaping. The combination of dynamic parameter adjustment and adaptive filtering produces a synergistic effect. Parameter adaptation ensures that the data stream sent to traditional devices has the highest quality potential at the digital level, while adaptive filtering ensures that this potential is faithfully reproduced to the greatest extent possible after the device's physical conversion. It avoids distortion caused by pushing audio data beyond the processing capacity of underperforming devices, while also preventing high-performance devices from failing to reach their full potential due to uncorrected signals. The final technical effect is that, regardless of the original performance of the traditional audio equipment, this method can significantly improve its final auditory performance, making its sound output more balanced, clear, and natural, achieving universal optimization and consistent improvement of sound quality across different devices.

[0059] In a specific embodiment of the present invention, step S6 includes: S6.1. For each conventional audio device, extract exclusive static audio parameters characterizing the upper limit of physical performance from the configuration file of each conventional audio device. At the same time, through the independent feedback loop corresponding to each conventional audio device in the bridging device, monitor the signal status of the analog output terminal of each conventional audio device in real time, and generate real-time audio quality feedback parameters exclusive to each conventional audio device. The exclusive static audio parameters include the maximum supported sampling rate and the maximum bit depth. The real-time audio quality feedback parameters include the signal-to-noise ratio and the frequency band distortion degree.

[0060] S6.2. For each traditional audio device, its unique static audio parameters and unique real-time audio quality feedback parameters are combined, and the device performance-transmission parameter two-dimensional lookup table stored in the data warehouse is called to determine the target bit depth adjustment parameters, bit rate adjustment parameters and sampling rate adjustment parameters for each traditional audio device.

[0061] It should be noted that the two-dimensional lookup table for device performance and transmission parameters is specifically constructed based on static performance test data and transmission adaptation experimental data of traditional audio devices.

[0062] S6.3. For each traditional audio device, the target bit depth adjustment parameters, bit rate adjustment parameters, and sampling rate adjustment parameters are combined to generate exclusive optimized transmission parameters for each traditional audio device, ensuring that the subsequent adaptation data, i.e., step S7, is fully compatible with the hardware characteristics and real-time operating status of each traditional audio device.

[0063] This method achieves refined and intelligent control of audio transmission parameters by combining the static theoretical performance of the device with its dynamic actual performance. Its technical advantage lies in not only setting parameters based on the device's potential but also making real-time adjustments based on the device's actual state. This synergistic mechanism, combining static characteristics with dynamic feedback, ensures that traditional audio devices are always provided with the optimal data stream under current conditions. It can match high-specification data streams to high-performance devices to fully leverage their sound quality advantages, while proactively downscaling devices with limited performance or bottlenecks when processing complex signals to ensure playback stability and distortion-free performance. Ultimately, this adaptive optimization strategy enables audio quality to reach an optimal balance across different devices and music content, surpassing simple combinations of techniques that rely solely on static profiles or passive distortion limitations.

[0064] In a specific embodiment of the present invention, the method for seamlessly bridging a conventional audio device to Auracast further includes the following step: S9. Obtaining a user device association connection instruction through a user interface.

[0065] S10. For each specified traditional audio device in the device association connection instruction, start a dedicated scanning process separately.

[0066] S11. In each traditional audio device, a different bridging process is implemented, outputting connection status feedback information corresponding to each traditional audio device. The bridging process includes scanning, identification, and connection.

[0067] In a specific embodiment of the present invention, step S11 includes: S11.1. For each traditional audio device, generate device-state associated connection state data for each traditional audio device according to the current processing flow state.

[0068] S11.2. For each traditional audio device, the device-state associated connection status data is converted into a dedicated feedback signal, which is a combination of a visual indication signal and a voice prompt signal.

[0069] S11.3. Drive the output device to emit the visual indication signal and the voice prompt signal.

[0070] Specifically, users can explicitly specify one or more traditional audio devices to be bridged by using physical buttons (such as long-pressing the button corresponding to the device identifier), touch-sensitive areas (such as clicking the device A icon displayed on the screen), or accompanying software applications (such as checking the device B / Device C to be connected). The bridging device parses the instruction and generates a device-instruction association table, such as instruction 1 corresponding to device A, instruction 2 corresponding to device B / C, rather than triggering an indiscriminate full device scan.

[0071] For each designated legacy audio device in the device-state associated connection status data, the bridging device sends a unique probe signal, such as a wideband test signal with a unique device identifier, to avoid interference with probe signals from other devices. Simultaneously, it only scans Auracast broadcast sources compatible with each designated legacy audio device. For example, based on the supported frequency band parameters of legacy audio device A, it filters out Auracast signals exceeding its supported frequency bands, generating a dedicated candidate broadcast source list for each legacy audio device, rather than generating a uniform global device list.

[0072] The bridging device assigns a unique feedback identifier to each traditional audio device. For example, traditional audio device A corresponds to a red indicator light and voice prompt for device A, while traditional audio device B corresponds to a blue indicator light and voice prompt for device B.

[0073] When traditional audio device A is in the Auracast broadcast source scanning stage, only its dedicated red indicator light flashes slowly; when traditional audio device B is in the traditional device characteristic identification stage, only its dedicated blue indicator light flashes quickly; when traditional audio device A is successfully connected, only its red indicator light is constantly on; if traditional audio device B fails to connect, its blue indicator light flashes. Users can directly judge the bridging progress of a single traditional audio device through the correspondence between the identifier and the status, rather than receiving a vague overall status signal.

[0074] For example, if a user wants to bridge both a vintage speaker A and wired headphones B simultaneously, they simply need to select A+B in the software. The bridging device will then send a probe signal labeled "A" to the vintage speaker A and a probe signal labeled "B" to the wired headphones B. During the scan, only Auracast sources matching the frequency band parameters supported by speaker A will be selected for the vintage speaker A, and Auracast sources matching the input impedance parameters of the wired headphones B will be selected for the wired headphones B. Feedback is provided by a green light indicating that the vintage speaker A is connected and a flashing yellow light indicating that the wired headphones B is being scanned, allowing the user to clearly distinguish the status of the two devices.

[0075] This approach achieves significant technological results by encapsulating complex technical processes behind an extremely simple user interaction model. It abstracts the multiple steps involved in traditional Bluetooth device pairing—such as searching, selecting, and confirming—into a single trigger action. This design greatly reduces the user's operational threshold and psychological burden, achieving a truly one-click, seamless bridging experience. The tight coupling between user connection commands and automated background processes, along with an intuitive status feedback mechanism, constitutes a complete, closed-loop interactive experience. This synergy is not merely a simple addition of functions, but rather creates a completely new and highly user-friendly experience, enabling even users unfamiliar with technology to effortlessly integrate older devices into the advanced Auracast ecosystem, thereby significantly improving the technology's universality and market acceptance.

[0076] In a specific embodiment of the present invention, the method for seamlessly bridging conventional audio devices with Auracast further includes the following steps: S12. For each conventional audio device, monitor the real-time audio stream quality parameters of the synchronous audio output individually, wherein the real-time audio stream quality parameters include total harmonic distortion plus noise and frame drop frequency.

[0077] It should be noted that the real-time audio stream quality parameters of the monitored synchronous audio output are specifically obtained by collecting the audio signals at the output end of the traditional audio devices through independent signal monitoring modules corresponding to the traditional audio devices in the bridging device, analyzing them separately, and generating real-time audio stream quality parameters for each traditional audio device.

[0078] S13. For each traditional audio device, based on the real-time audio stream quality parameters, individually call back and adjust the configuration file and buffer adjustment parameters of each traditional audio device.

[0079] It should be noted that when the total harmonic distortion plus noise parameter of a certain traditional audio device is... When the distortion exceeds the hardware tolerance threshold extracted from its initial configuration file, the distortion-equalizer adjustment rule table specific to this traditional audio device is invoked. This distortion-equalizer adjustment rule table is preset for the hardware characteristics of this traditional audio device and contains multiple sets of... Measured value range - corresponding adjustment parameter mapping relationship, for example: when When THD+Ni is in the 1% to 3% range, the corresponding gain reduction is 2dB in the 60Hz to 200Hz band and the output gain reduction is 1dB. When THD+Ni is in the 3% to 5% range, the corresponding gain reduction is 4dB in the 60Hz to 200Hz band, 1dB in the 2kHz to 5kHz band, and 2dB in the output gain, etc., depending on the current... The specific values ​​are determined by referring to a table, and the multi-band equalizer gain parameters and output gain parameters in the configuration file of the traditional audio device are adjusted to the target values ​​that match the distortion-equalizer adjustment rule table.

[0080] When the frame drop frequency parameter of a certain traditional audio device When the frame drop rate exceeds its preset frame drop threshold, the device's dedicated frame drop frequency-buffer parameter adjustment rule table is invoked. This rule table is preset based on the latency characteristics and network adaptability of the traditional audio device and contains multiple sets of... Frequency interval - corresponding α safety factor adjustment value mapping relationship, for example: when When the frequency is 1 to 3 times per minute, the corresponding α value is adjusted from the initial 1.2 to 1.5; when When the frequency is 4 to 6 times per minute, the corresponding α value is adjusted from the initial 1.2 to 1.8, etc., based on the current... The specific frequency is determined by referring to a table to determine the adjustment amount, and the corresponding buffer adjustment parameters for the traditional audio device are then applied. The α safety factor is adjusted to a target value that matches the frame drop frequency-buffer parameter adjustment rule table in order to quantitatively increase the buffer margin.

[0081] S14. Using the adjusted configuration files and buffer adjustment parameters for each traditional audio device, update steps S3 and S4 for each traditional audio device; to ensure that the core processing of each device can match its current sound quality status, achieving single-device optimization for single-device issues, rather than relying on a global unified update.

[0082] The technical principle of this invention lies in constructing a multi-stage, deeply coupled audio processing link from environmental perception to precise output. This method first actively scans to comprehensively acquire device information in the environment, forming a device list. The core innovation lies not only in device discovery but also in deep audio parameter identification of traditional audio devices based on the device list, generating personalized device profiles. These device profiles then guide the entire processing flow, directing targeted audio data conversion and ensuring that the output compatible audio data optimally matches the target device in terms of format and characteristics. Furthermore, a dynamic buffering adjustment mechanism is introduced to perform real-time flow control on the converted data stream, generating stable buffered audio data to adapt to uncertain transmission environments. Finally, through a synchronization control stage, the buffered data stream is precisely time-aligned, ultimately generating synchronized audio output that can be played consistently across multiple devices. The entire process forms an irreversible data processing flow, from macroscopic device discovery to microscopic parameter configuration, and then to dynamic flow control and timing synchronization, each step is interconnected, achieving intelligent bridging of traditional audio devices.

[0083] The technical effect of this invention is a significant improvement in the overall performance and user experience of traditional audio devices accessing Auracast broadcasting. Through automated device identification and configuration file generation, this method solves the problems of poor compatibility and cumbersome manual configuration required by existing technologies, achieving plug-and-play convenience. Precise conversion based on device configuration files ensures that the audio signal can fully realize the potential of traditional devices, avoiding sound quality loss or distortion that may result from universal conversion schemes, thereby greatly optimizing audio fidelity. The dynamic buffer adjustment step effectively overcomes latency and stuttering issues in audio transmission, ensuring high smoothness and stability in playback. The final synchronization control stage ensures perfect sound synchronization in multi-device playback scenarios, avoiding echoes or phase confusion, and creating a unified and harmonious sound field. In summary, this invention, through a series of collaborative steps, comprehensively solves the four major technical challenges of compatibility, sound quality, smoothness, and synchronization, enabling older audio devices to seamlessly and with high quality integrate into the modern audio broadcasting ecosystem.

[0084] It should be added that the formulas mentioned above, through the principle of dimensional consistency and mathematical standardization methods such as normalization, dimensionless parameter conversion, or unit system unification, can translate physical quantities with different properties into unitless standard values ​​or parameters that can be superimposed in the same dimension. This eliminates the interference of different dimensions on the computational logic, allowing the formulas to retain the original data distribution characteristics while possessing mathematical rationality and adaptability to objective laws. The above descriptions are merely exemplary embodiments of the present invention and should not be construed as limiting the scope of the invention.

[0085] It should also be noted that the various threshold settings described in this invention are based on long-term operational history data of traditional audio devices and the Auracast bridging system, audio transmission interruption and distortion failure cases, and extensive compatibility test results in multiple scenarios. At the same time, combined with the experience of experts in the field of audio transmission, the network environment fluctuation characteristics of different application scenarios, the design life and rated performance indicators of key hardware of traditional audio devices, and the human hearing perception threshold, the thresholds are calibrated and optimized in multiple rounds. This has a solid scientific basis and practical operability. Moreover, the relevant threshold calibration technology in the prior art is relatively mature, so it will not be elaborated here.

[0086] The above embodiments can be implemented, in whole or in part, by software, hardware, firmware, or any other combination thereof. When implemented in software, the above embodiments can be implemented, in whole or in part, as a computer program product. Those skilled in the art will recognize that the algorithm steps of the various examples described in conjunction with the embodiments disclosed herein can be implemented in electronic hardware, or a combination of computer software and electronic hardware. Whether these functions are implemented in hardware or software depends on the specific application and design constraints of the technical solution. Those skilled in the art can use different methods to implement the described functions for each specific application, but such implementation should not be considered beyond the scope of this application.

[0087] In addition, the functional modules in the various embodiments of this application can be integrated into one processing module, or each module can exist physically separately, or two or more modules can be integrated into one module.

[0088] The above description is merely a specific embodiment of this application, but the scope of protection of this application is not limited thereto. Any variations 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. Therefore, the scope of protection of this application should be determined by the scope of protection of the claims.

[0089] The above content is merely an example and illustration of the concept of the present invention. Those skilled in the art can make various modifications or additions to the specific embodiments described, or use similar methods to replace them, as long as they do not deviate from the concept of the invention or exceed the scope defined by the present invention, and all such modifications and additions should fall within the protection scope of the present invention.

Claims

1. A method for seamless bridging of Auracast for legacy audio devices, characterized in that, The method comprises: S1. Scan Auracast broadcast sources and traditional audio devices in the surrounding environment to generate a device list, the device list comprising identifiers, input impedance parameters, supported frequency band parameters, maximum bit depth, maximum supported sampling rate, device delay characteristic parameters, frequency response data, exclusive static audio parameters, hardware distortion tolerance threshold, multi-section equalizer gain parameters and output gain parameters of each traditional audio device; S2. Identify audio setting parameters of each traditional audio device according to the device list to generate a configuration file of each traditional audio device; S3. Convert Auracast audio data streams into compatible audio data of each traditional audio device based on the configuration file of each traditional audio device; S4. Perform dynamic buffer adjustment on filtered audio data obtained by performing exclusive digital equalization filtering and dynamic range compression correction on the compatible audio data of each traditional audio device to generate buffered audio data of each traditional audio device; S5. Apply synchronization control to the buffered audio data of each traditional audio device to generate synchronized audio output of all traditional audio devices.

2. The method of seamlessly bridging Auracast for legacy audio devices of claim 1, wherein, The S2 step comprises: S2.

1. Extract the identifiers, input impedance parameters and supported frequency band parameters of each traditional audio device from the device list; S2.

2. Import the input impedance parameter and the support frequency band parameter of each traditional audio device into the audio setting parameter model, output the matched audio setting parameter of each traditional audio device, and the audio setting parameter model is: ; wherein, a matching audio setting parameter of a generated first conventional audio device, an input impedance parameter of a first conventional audio device, a supported frequency band parameter of a first conventional audio device, a preset mapping rule set, the audio setting parameter is a set of adjustment instructions, r is a number of conventional audio devices, , a number of conventional audio devices, the audio setting parameter includes an output gain parameter, a multi-section equalizer gain parameter; S2.

3. Extract the maximum bit depth, maximum supported sampling rate, device delay characteristic parameters, frequency response data, exclusive static audio parameters and hardware distortion tolerance threshold of each traditional audio device from the device list, and combine the matching audio setting parameters, input impedance parameters and supported frequency band parameters of each traditional audio device to generate a configuration file of each traditional audio device, which is associated with and stored with the identifier of the corresponding traditional audio device.

3. The method of seamlessly bridging Auracast for legacy audio devices of claim 1, wherein, The S4 step comprises: S4.

1. For each traditional audio device, obtain network condition parameters of a network transmission link, and obtain device delay characteristic parameters from the configuration file of each traditional audio device, wherein the network condition parameters comprise a maximum network jitter value and a data packet loss rate monitored in a previous observation period, and the device delay characteristic parameters comprise a plurality of delay parameters inherent to each traditional audio device when processing audio data; S4.

2. For each legacy audio device, a dynamic buffer adjustment parameter is calculated using a buffer adjustment parameter calculation model based on the network condition parameter and the device latency characteristic parameter of the legacy audio device ; wherein, is a calculated dynamic buffer adjustment parameter of the first conventional audio device, i.e. a target buffer duration, is a comprehensive representation value obtained by processing the device delay characteristic parameter of the first conventional audio device read in the device profile, , are respectively a maximum network jitter value and a packet loss rate monitored in a previous observation period, is a dimensionless safety factor greater than 1, determined by a device delay-network jitter and dimensionless safety factor adaptation table, used to provide a buffer margin on the basis of the maximum jitter, is a preset gain function of packet loss rate and time. S4.

3. For the compatible audio data of each traditional audio device, apply the buffer adjustment parameters calculated therefor respectively, configure an independent first-in-first-out data buffer for processing, and obtain the filling amount of the data buffer of each traditional audio device in real time, dynamically adjust the reading speed of the compatible audio data of the corresponding data buffer according to the filling amount deviation of the data buffer of each traditional audio device from a target buffer duration, i.e. the filling amount deviation, in combination with the relationship between the filling amount deviation value stored in the data warehouse and the reading and writing speed, to generate buffered audio data of each traditional audio device.

4. The method of seamlessly bridging Auracast for legacy audio devices of claim 1, wherein, The S5 step comprises: S5.

1. Obtain the output timestamp parameter of each traditional audio device; S5.

2. From all the legacy audio devices, the first successfully connected legacy audio device is recorded as the master device, and the remaining legacy audio devices are recorded as several slave devices. For each slave device, its output timestamp parameter is compared with that of the master device, and the synchronization offset parameter of each slave device is calculated separately, and the calculation formula is ; wherein, is a synchronization offset parameter of the mth slave device, i.e. a playback rate adjustment factor, is a current master-slave clock offset of the mth slave device calculated after comparing the output timestamps of the mth slave device and the master device, is a dedicated integration value of the master-slave clock offset of the mth slave device accumulated over time, and are a proportional gain coefficient and an integral gain coefficient, respectively, is a number of legacy audio devices, , is a total number of legacy audio devices, is a subset of r, and ; S5.

3. For each slave device, use its exclusive synchronization offset parameter to individually correct the playback timing of the buffered audio data of each slave device, while maintaining the playback timing of the buffered audio data of the master device as a reference, to generate synchronized audio output of all traditional audio devices.

5. The method of seamlessly bridging Auracast from a legacy audio device according to claim 1, wherein, Further comprising the following steps: S6. For each legacy audio device, determine the exclusive optimized transmission parameters respectively; S7. For each legacy audio device, apply the exclusive optimized transmission parameters determined for it to perform parameter adaptation processing on the compatible audio data corresponding to each legacy audio device, and the parameter adaptation processing is specifically resampling or bit depth adjustment on the compatible audio data to generate exclusive adapted audio data for each legacy audio device; S8. For the exclusive adapted audio data of each legacy audio device, separately perform adaptive filtering processing: call the frequency response data in the configuration file of each legacy audio device to generate an exclusive digital equalization filter that is exactly opposite to the inherent frequency response curve of each legacy audio device, and at the same time, combine the real-time audio quality feedback parameters of each legacy audio device to enable an exclusive dynamic range compressor for each legacy audio device to generate exclusive filtered audio data for each legacy audio device, and use the exclusive filtered audio data of each legacy audio device as the input of the corresponding S4 step for each legacy audio device.

6. The method of seamlessly bridging Auracast from a legacy audio device according to claim 5, wherein, The S6 step comprises: S6.

1. For each legacy audio device, extract the exclusive static audio parameters representing the physical performance upper limit from the configuration file of each legacy audio device, and at the same time, monitor the signal state of the analog output end of each legacy audio device through the independent feedback loop in the bridging device corresponding to each legacy audio device to generate real-time audio quality feedback parameters exclusive to each legacy audio device, and the exclusive static audio parameters include maximum supported sampling rate and highest bit depth, and the real-time audio quality feedback parameters include signal-to-noise ratio and frequency band distortion degree; S6.

2. For each legacy audio device, respectively integrate its exclusive static audio parameters, exclusive real-time audio quality feedback parameters, and call the device performance-transmission parameter two-dimensional lookup table stored in the data warehouse to determine the target bit depth adjustment parameter, bit rate adjustment parameter and sampling rate adjustment parameter of each legacy audio device respectively; S6.

3. For each legacy audio device, combine the target bit depth adjustment parameter, bit rate adjustment parameter and sampling rate adjustment parameter determined for it to generate the exclusive optimized transmission parameters of each legacy audio device.

7. The method of seamlessly bridging Auracast from a legacy audio device of claim 1, wherein, Further comprising the following steps: S9. Obtain the user device association connection instruction through the user interface; S10. For each legacy audio device specified in the device association connection instruction, separately start an exclusive scanning process; S11. Output the connection state feedback information corresponding to each legacy audio device in the different bridging processes of each legacy audio device, and the bridging process includes scanning, identification and connection.

8. The method of seamlessly bridging Auracast from a legacy audio device according to claim 7, wherein, The S11 step comprises: S11.

1. For each legacy audio device, generate device-state associated connection state data of each legacy audio device according to the state of the current processing process; S11.

2. For the device-state associated connection state data of each legacy audio device, separately convert it into an exclusive feedback signal, and the exclusive feedback signal is a combination of visual indication signal and voice prompt signal; S11.

3. The driving output device sends out the visual indication signal and the voice prompt signal.

9. The method of seamlessly bridging Auracast from a legacy audio device of claim 1, wherein, Further comprising the following steps: S12. For each legacy audio device, separately monitor real-time audio stream quality parameters of the synchronized audio output, the real-time audio stream quality parameters including total harmonic distortion plus noise, frame drop frequency; S13. For each legacy audio device, based on the real-time audio stream quality parameters, separately call back to adjust the configuration file and buffer adjustment parameters of each legacy audio device; S14. Update the corresponding S3 step and S4 step of each legacy audio device using the adjusted configuration file and buffer adjustment parameters of each legacy audio device.

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