Wireless microphone receiver with de-voice function

Abstract

This invention relates to the field of audio processing equipment technology, and discloses a wireless microphone receiver with acoustic cancellation function, including a shell, external interaction components, and an internal circuit system. The external interaction components include an interface, indicator lights, and a power button. The circuit system includes a power supply module, a wireless communication receiving module, and a core processing cluster. The core processing cluster includes an audio effects processing DSP and an acoustic cancellation DSP. The power button is connected to the enable terminal of a step-down circuit and the pins of the audio effects processing DSP. The audio effects processing DSP connects to the Bluetooth unit and the microphone receiving module, and establishes a cross-chip data link with the acoustic cancellation DSP. This invention adopts a dual-processor architecture, using the power button to control the closure of the underlying power supply and switch audio routing modes. In acoustic cancellation mode, the acoustic cancellation DSP separates frequency bands to filter out the original vocals and extract independent accompaniment, while the audio effects processing DSP synchronously mixes in the microphone vocal sequence and converts it into analog audio output, realizing multi-source audio reception and integrated mixing.

Description

Technical Field

[0001] This invention relates to the field of audio processing equipment technology, and in particular to a wireless microphone receiver with acoustic cancellation function. Background Technology

[0002] Existing portable entertainment singing devices have limitations in processing audio sources. Currently common audio processing devices rely on dedicated application software to provide pure accompaniment audio. When users directly use standard audio files from digital music platforms (i.e., composite audio containing the original vocals) as external sound sources, the lack of audio separation and original sound removal mechanisms in the receiving device causes the user's voice picked up by the microphone to overlap and mix with the original vocals from the external sound source. This results in a lack of layering in the final output synthesized audio, failing to meet the user's need for mixing and singing directly from ordinary music files.

[0003] Meanwhile, with the popularization of digital music copyrights and the segmentation of platform memberships, traditional fixed entertainment venues face the problem of limited music library content, and the fixed equipment environment is difficult to adapt to users' usage preferences in scenarios such as home, car, or outdoors. In order to perform at any time using the dedicated music library of personal terminals (such as mobile phones, smart TVs, or in-car entertainment systems), existing equipment setup solutions require combining separate Bluetooth audio receivers, wireless microphone receivers, and external mixers. This separate hardware structure requires signal bridging through multiple cables, resulting in cumbersome equipment connection layers.

[0004] Furthermore, due to the dispersed functions of modular devices, numerous independent physical buttons or toggle switches are required for operations such as power control, Bluetooth pairing, signal channel switching, and operating mode changes. This complex external interaction structure not only increases the redundancy of the hardware design but also complicates the user's workflow. If attempting to integrate multi-channel audio reception and vocal cancellation functions into a single portable device, the traditional single-processor chip architecture is highly susceptible to limitations in computing power allocation when simultaneously handling high-load frequency band separation, vocal feature filtering, and multi-source data synchronization. This can easily lead to incomplete vocal cancellation or distortion of the accompaniment frequency band, making it difficult to guarantee the final quality of the mixed output. Summary of the Invention

[0005] The purpose of this invention is to provide a wireless microphone receiver with acoustic removal function, which aims to solve the problems of cumbersome setup and complex cable connections of external mixing equipment, redundant function buttons leading to complicated operation, and the difficulty of performing high-quality acoustic removal and accompaniment extraction on ordinary audio with original vocals in the existing single-chip architecture.

[0006] To address the aforementioned technical problems, this invention provides a wireless microphone receiver with acoustic cancellation functionality, comprising a housing, external interaction components housed within the housing, and a circuit system housed within the housing. The external interaction components include a USB charging port, a 3.5mm analog interface, an LED indicator, and a power button. The circuit system includes a power supply and control module. The charging circuit within the power supply and control module connects the USB charging port to a lithium battery assembly. The lithium battery assembly is connected to a DC-DC step-down power supply module, which converts the output voltage of the lithium battery assembly into the 3.3V DC operating voltage required by the subsequent chips. The power button is synchronously connected to the enable pin of the DC-DC step-down power supply module and the general-purpose input pin of the audio processing DSP in the core processing cluster included in the circuit system. The audio processing DSP is connected to the LED indicator, the 3.5mm analog interface, and the Bluetooth front-end unit in the wireless communication and receiving module included in the circuit system. The microphone receiving module in the wireless communication and receiving module is connected to the audio processing DSP via an I2S bus. The audio processing DSP and the acoustic cancellation DSP in the core processing cluster establish a cross-chip data link via a bidirectional I2S audio bus and a UART communication bus.

[0007] This technical solution separates sound effect processing and original sound removal operations through a dual DSP architecture. It uses dual control signals generated by the power button to synchronously trigger the step-down circuit and processor pins to achieve power supply control and system startup.

[0008] Preferably, when the duration of pressing the power button exceeds a set first time threshold, the power button closes the underlying circuit and simultaneously generates dual control signals. One signal branch is transmitted to the enable pin of the DC-DC step-down power supply module, and the other signal branch is transmitted to the general-purpose input pin of the audio processing DSP. After the microcontroller unit inside the audio processing DSP is connected to the operating voltage, it executes a power-on reset sequence, loads the boot firmware from the internal non-volatile memory, sets the system's phase-locked loop clock tree, and initializes the communication pins of each data bus to a high-impedance state. The audio processing DSP controls the LED indicator drive pin to output a high level, driving the LED indicator to be continuously illuminated. The audio processing DSP uses its internally integrated analog-to-digital converter to acquire the current terminal output voltage of the lithium battery component. The first time threshold is a preset time constant in the internal non-volatile memory, used to characterize the duration required for the power button to close the underlying circuit.

[0009] Furthermore, the audio processing DSP sends address bytes to the connected microphone receiving module via the I2C communication bus and writes multiple sets of operating parameters to the control register. These operating parameters specify the sampling rate and bit depth of the I2S bus between the microphone receiving module and the audio processing DSP, and set the initial wireless frequency band for the RF front-end circuit. The audio processing DSP sends an initialization command in data frame format to the sound cancellation DSP via the UART communication bus. Upon receiving the command, the sound cancellation DSP clears its internal audio data buffer and resets the bidirectional I2S audio bus transceiver status connected to the audio processing DSP in a closed loop.

[0010] In one specific embodiment, the Bluetooth front-end unit establishes a wireless connection with an external audio source device and receives a digital composite audio stream containing vocal and accompaniment information. The audio effects processing DSP allocates a circular buffer in its internal random access memory and writes the digital composite audio stream into the circular buffer. The microphone receiving module drives an external RF antenna to scan the spatial channel, receives the RF vocal signal emitted by an independent wireless microphone, and down-converts the received RF vocal signal to baseband. The analog-to-digital converter unit inside the audio effects processing DSP resamples the baseband signal and demodulates the digital vocal sequence containing the user's voice characteristics.

[0011] Furthermore, the audio effects processing DSP calls its internally embedded digital signal processing algorithms to perform noise reduction, equalization, and spatial reverberation operations on the digital vocal sequence, generating a processed digital vocal sequence. The audio effects processing DSP uses an internal hardware timer to compare the generation timestamp of the processed digital vocal sequence with the arrival timestamp of the digital composite audio stream in the circular buffer, performing synchronization alignment operations such as silencing, zero padding, or discarding old data frames.

[0012] Synchronization alignment is performed by comparing timestamps to eliminate data delay differences between the Bluetooth link and the RF link, thus maintaining audio phase consistency.

[0013] Preferably, the microcontroller inside the audio processing DSP continuously monitors the level status of the general-purpose input pin connected to the power button. The microcontroller toggles the routing status flag set in the internal non-volatile memory. When the routing status flag is at a logic low level, it is determined that the audio is in native mode. The microcontroller controls the internal audio bus to directly connect the external Bluetooth channel and the internal mixing calculation unit. The digital composite audio stream, stored in the circular buffer and having undergone synchronization alignment, is directly sent to the multiply-accumulate unit inside the audio processing DSP.

[0014] When the routing status flag is at a logic high level, it is determined that the original sound cancellation mode has been entered. The audio processing DSP changes the internal audio bus topology, disconnecting the data pass-through link between the ring buffer and the internal multiply-accumulate unit. The audio processing DSP sends the digital composite audio stream to the original sound cancellation DSP through the bidirectional I2S audio bus. The data parsing module inside the original sound cancellation DSP decomposes the multiplexed data frame into independent left and right channel audio sequences. The digital crossover integrated inside the original sound cancellation DSP divides the separated channel data into three independent frequency band branches: low frequency, mid frequency, and high frequency. The original sound cancellation DSP performs energy normalization on the left and right channel sequences in the mid-frequency band, filters out the vocal frequency band components, generates an independent digital accompaniment sequence, and stores it in the internal transmit buffer.

[0015] Switching between original sound mode and original sound removal mode is achieved by monitoring the button level. In original sound removal mode, a digital crossover is used to separate frequency bands, filtering out mid-frequency vocal signals while retaining low-frequency and high-frequency accompaniment components.

[0016] Furthermore, the original sound DSP allocates a second set of general-purpose input / output pins and multiplexes them as an I2S master control transmit bus, sending the independent digital accompaniment sequence in the transmit buffer back to the audio effects processing DSP. The audio effects processing DSP opens a first-in-first-out (FIFO) delay queue in its main memory. The multiply-accumulate unit inside the audio effects processing DSP synchronously extracts the independent digital accompaniment sequence and the processed digital vocal sequence after delay alignment operations performed by the FIFO delay queue, and adds them together to generate the final digital mix sequence. The FIFO delay queue compensates for the delay caused by cross-chip transmission and processing operations, achieving data synchronization between the accompaniment sequence and the processed digital vocal sequence.

[0017] In one specific embodiment, the DAC (Digital-to-Analog Converter) integrated within the audio processing DSP receives the final digital mix sequence and maps it into a continuous-time analog audio signal. The DAC, relying on an internal resistor network or switched capacitor array, converts discrete digital quantization values ​​into corresponding stepped voltage distributions. An analog low-pass reconstruction filter is connected in series at the back end of the DAC to generate the continuous-time analog audio signal. This continuous-time analog audio signal is transmitted via internal traces on the printed circuit board to a 3.5mm analog interface on the receiver housing. A DC-blocking coupling capacitor is placed between the signal traces and the 3.5mm analog interface on the printed circuit board.

[0018] Preferably, the 3.5mm analog interface is used to adapt to an external 3.5mm analog audio cable. The 3.5mm analog interface connects to an independent external speaker via the 3.5mm analog audio cable, allowing the continuous-time analog audio signal to flow unidirectionally into the independent external speaker for sound production; alternatively, the 3.5mm analog interface connects to the auxiliary analog input interface of a smart TV or in-vehicle entertainment system via the 3.5mm analog audio cable, transmitting the continuous-time analog audio signal back to the original audio source device for playback.

[0019] In summary, the present invention has at least one of the following beneficial technical effects:

[0020] 1. This invention integrates a Bluetooth front-end unit and a microphone receiving module to simultaneously acquire digital composite audio streams transmitted from external mobile phones, smart TVs, or in-vehicle terminals, as well as radio frequency voice signals transmitted from a wireless microphone. After processing by the internal circuitry and DAC digital-to-analog conversion, the two wireless signals are output to external speakers or in-vehicle auxiliary input interfaces via a unified 3.5mm analog interface. This reduces the number of cable connection layers between audio devices, enabling users to quickly establish audio mixing links in home, in-vehicle, or outdoor settings.

[0021] 2. This invention achieves hardware function reuse for interactive buttons by simultaneously connecting the individual power button to the enable terminal of the DC-DC step-down power supply module and the general-purpose input pin of the audio processing DSP. The microcontroller inside the audio processing DSP determines whether to close the underlying circuitry, trigger RF channel scanning and pairing, or initialize the parameters of each module via the I2C and UART communication buses based on the duration of the press. Simultaneously, by capturing the short-press pulse and toggling the routing status flag, the device can directly switch between original sound mode and original sound-cancelled mode, simplifying the device's external structure and user operation process.

[0022] 3. This invention employs a dual-processor architecture of an audio effects processing DSP and a noise reduction DSP to handle complex audio separation and synthesis tasks. In noise reduction mode, the audio effects processing DSP cuts off the direct data link and redirects the composite audio stream to the noise reduction DSP. The noise reduction DSP uses a digital crossover to separate frequency bands and filter out the original vocals, extracting an independent digital accompaniment sequence before sending it back. The multiply-accumulate unit inside the audio effects processing DSP then adds the digital accompaniment sequence to the noise-reduced microphone digital vocal sequence, thus distributing the audio computation load. While eliminating the original vocal signal, it extracts an independent accompaniment, ensuring the output quality after the vocals and accompaniment are mixed. Attached Figure Description

[0023] Figure 1 This is a schematic diagram of the structure of the original sound-canceling receiver of the present invention;

[0024] Figure 2This is a schematic diagram of the 3.5mm analog interface of the receiver of the present invention;

[0025] Figure 3 This is a schematic diagram of an application scenario of the original sound cancellation function receiver of the present invention;

[0026] Figure 4 This is a schematic diagram of the second application scenario of the original sound cancellation function receiver of the present invention;

[0027] Figure 5 This is a hardware block diagram of the original sound cancellation function receiver of the present invention;

[0028] Figure 6 This is the original sound elimination audio path diagram of the present invention;

[0029] Figure 7 This is a frequency domain comparison diagram before and after DSP processing for removing the original sound in a specific application embodiment of the present invention.

[0030] The components include: 1. Outer shell; 2. USB charging port; 3. LED indicator; 4. Power button; 5. 3.5mm analog interface. Detailed Implementation

[0031] The following is in conjunction with the appendix Figure 1 -Appendix Figure 7 The present invention will be further described in detail below.

[0032] The present invention provides a wireless microphone receiver with original sound cancellation function, which may include: a housing 1, an external interaction component disposed on the surface of the housing 1, and a circuit system housed inside the housing 1.

[0033] The receiver's housing 1 features a rounded frame design. External interaction components are distributed on the surface of housing 1, including a USB charging port 2, a 3.5mm analog jack 5, an LED indicator 3, and a power button 4. The USB charging port 2 is located on the side for connecting to an external DC power supply. The 3.5mm analog jack 5 is located at one end of housing 1 for outputting analog audio signals, and its base has a fixing structure for mounting an adapter. The LED indicator 3 and power button 4 are located side-by-side at the other end of housing 1. The LED indicator 3 uses different colored flashing to indicate the device status, while the power button 4 receives user-triggered press commands.

[0034] The circuit system housed within the casing 1 includes a wireless communication and receiving module, a core processing cluster, and a power supply and control module. The wireless communication and receiving module includes a Bluetooth front-end unit and a microphone receiving module. The core processing cluster employs a dual-processor physically isolated architecture, including an audio processing DSP and a sound cancellation DSP. The power supply and control module includes a charging circuit, a lithium battery assembly with a protection board, and a DC-DC step-down power supply module.

[0035] The system has corresponding hardware interconnections. The external USB charging port 2 is connected to the charging circuit, which is connected to the lithium battery pack. The DC-DC step-down power supply module converts the output voltage of the lithium battery pack into the 3.3V DC operating voltage required by the subsequent chips. The power button 4 located on the end face of the casing 1 is simultaneously connected to the enable terminal of the DC-DC step-down power supply module and the general-purpose input pin of the audio processing DSP. The audio processing DSP is connected to the LED indicator 3 and the 3.5mm analog interface 5, respectively. The microphone receiving module is connected to the audio processing DSP via the I2S bus (including the I2S receive pin and transmit pin). The audio processing DSP and the original sound cancellation DSP establish a cross-chip data link through the bidirectional I2S audio bus and the UART communication bus.

[0036] See attached document Figure 1 -Appendix Figure 6 This invention provides an audio processing method for a wireless microphone receiver with acoustic cancellation function, comprising the following steps:

[0037] S10, Perform system initialization: Press and hold the power button 4 to close the power supply circuit, start the microcontroller inside the sound processing DSP and execute the initialization program, drive the LED indicator 3 to light up, set the working frequency band and pin status of the microphone receiving module through the I2C bus, and reset the Bluetooth front-end unit and the original sound cancellation DSP.

[0038] S20, establish a communication link and receive audio signals: the Bluetooth front-end unit establishes a wireless connection with the external audio source device and receives a digital composite audio stream containing human voice and accompaniment information. The external audio source device is a mobile phone, TV or car stereo. The microphone receiving module acquires the human voice radio frequency signal, demodulates it into a digital human voice sequence and transmits it to the audio processing DSP. The audio processing DSP performs noise reduction and spatial reverberation processing on the digital human voice sequence.

[0039] S30, Perform audio routing and mixing: Short press the power button 4 to switch audio routing modes; In the original mode, the audio processing DSP directly adds the processed digital vocal sequence to the digital composite audio stream to complete the digital mixing; In the original mode, the audio processing DSP sends the digital composite audio stream to the original mode DSP via the I2S bus. The original mode DSP filters out the vocal frequency band components to generate an independent digital accompaniment sequence, and sends the digital accompaniment sequence back to the audio processing DSP via the I2S bus. The audio processing DSP adds the returned digital accompaniment sequence to the processed digital vocal sequence to complete the digital mixing;

[0040] S40 performs digital-to-analog conversion and output: The DAC digital-to-analog conversion unit integrated inside the audio processing DSP receives the digital mixing sequence, maps the digital mixing sequence into a continuous-time analog audio signal, and transmits it to the 3.5mm analog interface 5 for output to an external audio playback device. The external audio playback device is an independent external speaker or the analog input interface of a TV or car stereo.

[0041] The above steps will be described in detail below with reference to specific embodiments. Regarding the system initialization step, the process relies on the internal power supply circuit and communication bus topology of the device. Specifically, the system initialization step S10 includes the following sub-steps:

[0042] S11, close the power supply circuit and start the main control core. As a preferred method, the device's power supply state is controlled by the user's continuous pressing of the power button 4 on the surface of the casing 1. When the duration of the pressing action exceeds a set first time threshold (a preset time constant in the internal non-volatile memory, representing the duration required for the power button 4 to close the underlying circuit), the power button 4 closes the underlying circuit and simultaneously generates dual control signals. One signal branch is transmitted to the enable pin of the DC-DC step-down power supply module, and the other signal branch is transmitted to the general-purpose input pin of the audio processing DSP.

[0043] In this embodiment, the first time threshold is configured to range from 2 seconds to 4 seconds, preferably 3 seconds. After detecting the enable signal, the DC-DC step-down power supply module turns on its internal power switching circuit, converting the fluctuating DC voltage stored in a single lithium battery into a stable 3.3V DC operating voltage, and synchronously delivering the stepped-down power to the microphone receiving module and the audio processing DSP. After the microcontroller unit inside the audio processing DSP receives the operating voltage, it executes a power-on reset sequence, loads the boot firmware from its internal non-volatile memory, configures the system's phase-locked loop clock tree, and initializes the communication pins of each data bus to a high-impedance state.

[0044] S12, establish status feedback and perform battery power assessment. After the audio processing DSP completes the basic clock configuration, it controls the directly connected white LED indicator driver pin to output a high level, driving the white LED indicator to be continuously lit, indicating the device is powered on. Simultaneously, the audio processing DSP uses its internally integrated analog-to-digital converter to collect the current terminal output voltage of the lithium battery. Considering that the battery output is prone to transient voltage drops due to load changes, the audio processing DSP performs multiple continuous samplings of the voltage channel and calculates a moving average. The collected average value is then compared with a preset safe discharge voltage threshold. In this embodiment, the safe discharge voltage threshold is set in the range of 3.1V to 3.3V. If the comparison result determines that the collected average value is lower than the aforementioned discharge voltage threshold, the audio processing DSP cuts off the white LED indicator driver pin and outputs a pulse square wave with a specific duty cycle to the connected red LED indicator, implementing a low battery warning through flashing at a set frequency. The pulse square wave duty cycle can be selected as 50%, and the flashing frequency can be selected as the range of 1Hz to 2Hz.

[0045] S13, Configure the spatial RF band and bus communication status. The audio processing DSP sends address bytes to the connected microphone receiving module via the I2C communication bus and writes multiple sets of operating parameters to the control register. The written parameters specify the sampling rate and bit depth of the I2S audio bus between the two and set the initial wireless band for the RF front-end circuit. In this embodiment, the initial frequency band parameters sent by the audio processing DSP to the microphone receiving module are determined by the RF channel center frequency calculation formula:

[0046] ;

[0047] In the formula, Indicates the set center frequency of the radio frequency channel; This indicates the reference frequency provided by the internal crystal oscillator circuit of the microphone receiver module, typically 16MHz or 24MHz. This represents the integer frequency division coefficient configured inside the phase-locked loop module, and its value is a positive integer; This represents the fractional frequency division coefficient configured internally within the phase-locked loop module; This represents the fixed modulus value of the fractional frequency divider, and is limited by the underlying algorithm logic of the system. To prevent division overflow exceptions, the non-zero value must be equal to zero. This represents the overall synthesized frequency division ratio parameter that controls the output of the radio frequency voltage-controlled oscillator.

[0048] After sending the microphone receiving module parameters, the audio processing DSP performs a software reset on its connected Bluetooth antenna RF circuit and internal Bluetooth protocol stack. Simultaneously, the audio processing DSP sends an initialization command for the data frame format to the acoustic cancellation DSP via the UART control bus. Upon receiving the command, the acoustic cancellation DSP clears its internal audio data buffer and resets the bidirectional I2S audio bus transmit / receive state connected to the audio processing DSP in a closed loop. After executing these commands, the Bluetooth receiving unit, microphone receiving module, and acoustic cancellation DSP enter standby mode.

[0049] After the system hardware resources have completed initialization and are in standby mode, the device enters the space communication configuration stage, executing step S20 to establish a communication link and receive audio signals, specifically including the following processing sub-steps:

[0050] S21, establish a Bluetooth audio transmission link and acquire composite audio. The Bluetooth protocol stack integrated within the audio processing DSP controls the connected Bluetooth antenna to listen for wireless broadcast signals in the space. When the user initiates a connection request on an external audio source device, the audio processing DSP and the external audio source device complete key exchange and handshake pairing. In this embodiment, the external audio source device includes a mobile phone, smart TV, or in-vehicle terminal equipped with a Bluetooth transmission module. After pairing, the external audio source device encodes the audio data stream containing vocals and accompaniment information into radio frequency data packets for transmission. The Bluetooth antenna captures the aforementioned radio frequency data packets, and the baseband processing unit within the audio processing DSP performs unpacking and audio decoding operations on the received data packets to generate a discrete digital composite audio stream. To compensate for Bluetooth over-the-air transmission protocol latency, the audio processing DSP opens an independent, contiguous circular buffer in its internal random access memory, writes the aforementioned digital composite audio stream into the circular buffer, and uses it as a reference time base for subsequent mixing alignment.

[0051] While writing the digital composite audio stream into the circular buffer, the system executes S22 in parallel to establish an RF voice link and demodulate the digital voice sequence. Preferably, the device pairing state is triggered by the user pressing the power button 4. When the device is powered off and the power button 4 is pressed for an extended period exceeding a set second time threshold, the RF search mechanism is triggered. In this embodiment, the second time threshold is configured to range from 8 to 12 seconds, preferably set to 10 seconds. After the triggering action occurs, the audio processing DSP drives the white LED indicator to enter a rapid flashing state and sends a channel scanning command to the microphone receiving module. The microphone receiving module, based on the operating frequency band set in the aforementioned steps, drives the external RF antenna to scan the spatial channel and receive the RF voice signal emitted by the independent wireless microphone. The RF front-end circuit inside the microphone receiving module down-converts the received RF voice signal to baseband, and the analog-to-digital converter inside the audio processing DSP resamples the baseband signal, demodulating the digital voice sequence containing the user's voice characteristics. After demodulation, the microphone receiving module encapsulates the digital voice sequence into standard I2S format data according to the set clock rate and transmits it to the input pin of the audio processing DSP.

[0052] With data input, the system enters S23, performing spatial reverberation and noise reduction processing. After receiving the digital vocal sequence transmitted from the microphone receiver module, the audio effects processing DSP calls its internally built-in digital signal processing algorithm to perform noise reduction, equalization, and spatial reverberation operations on the input sequence, generating the processed digital vocal sequence. The audio effects processing DSP filters out out-of-band noise in the low and high frequency bands using its internal digital filters and introduces delay and gain attenuation components on the time axis. The vocal signal modification algorithm within the audio effects processing DSP is calculated based on the vocal signal modification processing formula:

[0053] ;

[0054] In the formula, Indicates the index number of the current sampling point in the discrete-time series; This represents the current sample value of the digital human voice sequence output after algorithmic processing; This represents the current sample value of the raw digital human voice sequence input to the microphone receiving module; The direct gain coefficient represents the main acoustic signal, and its value range is configured to be from 0.8 to 1.2. This represents the total number of spatial reverberation reflection paths set, and its value is a positive integer. Indicates the first The acoustic attenuation coefficient of each reflection path is set to an open interval value greater than 0 and less than 1, which is used to characterize the energy loss of sound waves after colliding with the wall. Indicates the first The number of delay sample points for each reflection path compared to the direct sound is calculated by multiplying the preset physical delay time by the system audio sampling rate; This represents the overall superposition response parameter of multipath delay reverberation; This represents the estimated environmental background noise sequence sample value, which is obtained by the audio processing DSP through continuous tracking and recording of the environmental background noise spectrum during periods without speech activity. This represents the dynamic noise reduction depth coefficient, with a value ranging from 0.1 to 0.5.

[0055] Through the aforementioned algorithm, the audio processing DSP converts the original input signal into a processed digital vocal sequence. To prevent sampling misalignment during subsequent merging, the audio processing DSP uses an internal hardware timer to compare the generation timestamp of the processed digital vocal sequence with the arrival timestamp of the digital composite audio stream in the aforementioned circular buffer, performing synchronization alignment operations such as silencing, zero-padding, or discarding outdated data frames. At this point, the audio processing DSP internally caches the time-synchronized digital composite audio stream and the digital vocal sequence for subsequent audio routing and mixing processing.

[0056] After establishing the dual-end data synchronization baseline, the device proceeds to the audio routing and mixing step S30. In this embodiment, the audio data routing is dynamically allocated by the operation commands, specifically encompassing the following processing sub-steps:

[0057] S31 captures button pulse commands and switches the audio routing matrix. The microcontroller inside the audio effects processing DSP continuously monitors the level status of the general-purpose input pin connected to power button 4. When the general-purpose input pin captures a transient falling edge signal from high to low, the microcontroller starts an internal hardware timer for timing. When the contact opens and generates a rising edge signal, the microcontroller reads the time count value recorded by the hardware timer and calculates the pulse duration. The formula for determining the pulse duration is:

[0058] ;

[0059] In the formula, This indicates the calculated actual compression pulse duration; This represents the discrete count value accumulated by the microcontroller's internal hardware timer between the falling and rising edges, and the value is a non-negative integer. This indicates the operating clock frequency provided to the internal hardware timer, set at the system level. A non-zero positive value is used to prevent division overflow exceptions.

[0060] As a preferred approach, the system sets the mechanical vibration filtering threshold to 50 milliseconds and the short press judgment upper limit threshold to 1.5 seconds. If the actual press pulse duration is less than the mechanical vibration filtering threshold, the microcontroller determines that the current level change is an interference glitch caused by the vibration of the button spring and discards it. If the actual press pulse duration is between the mechanical vibration filtering threshold and the short press judgment upper limit threshold, the microcontroller confirms that it has received a compliant short press pulse signal and excludes interference from long press power-on commands exceeding 2 seconds.

[0061] Upon receiving the short-press pulse signal, the microcontroller toggles the routing status flag in its internal non-volatile memory. When the status flag is at a logic low level, the system determines it is in native mode, and the microcontroller controls the internal audio bus to close the direct connection between the external Bluetooth channel and the internal mixing unit. When the status flag is at a logic high level, the system determines it is in native mode, the microcontroller changes the internal audio bus topology, cuts off the aforementioned direct connection, and redirects the data from the Bluetooth audio channel to the I2S transmit pin connected to the native mode DSP.

[0062] S32 performs data pass-through and overlay mixing in acoustic mode. When the status flag is detected to be at a low logic level, the system operates in acoustic mode. The digital composite audio stream, stored in a circular buffer and already synchronized, is directly fed into the multiply-accumulate unit within the audio processing DSP. The data input to the multiply-accumulate unit is a processed digital vocal sequence from the microphone link. The audio processing DSP uses a weighted attenuation mechanism to scale and blend the two input sequences before addition. The acoustic digital mixing calculation formula within the audio processing DSP is as follows:

[0063] ;

[0064] In the formula, Indicates the index number of the current sampling point in the discrete-time series; This represents the current sample value of the digital mix sequence directly generated through superposition calculation; This represents the current sample value of the digital composite audio stream read from the circular buffer; This represents the current sample value of the processed digital human voice sequence output from the preceding steps; This represents the scaling weight factor for the external digital composite audio stream channel, with a value range of 0.5 to 0.8. This represents the scaling weight coefficient for the microphone's vocal channel, with a value range of 0.7 to 1.0. Combined with the hardware limiter within the audio processing DSP to limit the total amplitude, it generates the final digital mix sequence.

[0065] When the status flag is detected to be at a logic high level, the system enters the original sound cancellation mode and executes processing sub-step S33, which performs path blocking and data redirection in the original sound cancellation mode.

[0066] In this mode, the audio processing DSP changes the node connectivity of its internal digital audio cross-matrix, disconnecting the data pass-through between the circular buffer and the internal multiply-accumulate unit. The microcontroller calls the integrated direct memory access controller to periodically transfer the digital composite audio stream temporarily stored in the circular buffer to the I2S transmit data register inside the audio processing DSP. The frequency divider circuit inside the audio processing DSP generates a synchronized bit clock and channel selection signal based on the system master clock. The formula for calculating the bus bit clock frequency is as follows:

[0067] ;

[0068] In the formula, This indicates the clock frequency of the I2S bus bit output to the original sound DSP; The system audio sampling rate of the digital composite audio stream is specified; the configuration option is selected as 44.1kHz or 48kHz. This indicates the quantization bit width of a single audio sample, and can be set to 16 or 24. This indicates the total number of channels contained in the audio data. In this embodiment, it is set to 2, representing a stereo two-channel transmission format.

[0069] Upon receiving the external audio stream, the system executes processing sub-step S34, filtering out vocal frequency band features and extracting the accompaniment based on hardware-level acoustic algorithms. The data parsing module within the de-sound DSP pre-decomposes the multiplexed data frames into independent left and right channel audio sequences. The integrated digital crossover frequency divider within the de-sound DSP divides the separated channel data into three independent frequency bands: low, mid, and high frequencies, based on a preset cutoff frequency. In this embodiment, the lower threshold of the vocal frequency band is preferably configured as 200Hz; the upper threshold is preferably configured as 4000Hz. Before frequency band cancellation, the de-sound DSP performs energy normalization on the left and right channel sequences in the mid-frequency band. The accompaniment extraction calculation formula within the de-sound DSP is as follows:

[0070] ;

[0071] In the formula, Indicates the index number of the current sampling point in the discrete-time series; This represents the current sample value of the extracted independent digital accompaniment sequence; This represents the low-pass sample value after combining the low-frequency bands of the left and right channels, used to preserve the energy of drum beats and bass guitar. This represents the high-pass sample value after merging the high-frequency bands of the left and right channels, used to preserve cymbal and spatial overtone details; This represents the current sample value output by the left channel sequence after passing through the mid-frequency bandpass filter; This represents the current sample value output by the right channel sequence after passing through the mid-frequency bandpass filter; This represents the low-frequency retention factor, with a value range of 0.8 to 1.0. This represents the high-frequency retention factor, with a value range of 0.8 to 1.0. This represents the mid-frequency cancellation gain coefficient, with a value range of 1.0 to 1.5, used to compensate for the overall sound field energy attenuation caused by the subtraction operation; This indicates the central image cancellation term, which uses phase reversal characteristics to eliminate human voice components with similar amplitude and phase in the left and right channels.

[0072] After processing, the composite audio stream containing the original vocal features is converted into an independent digital accompaniment sequence with the mid-frequency center image removed.

[0073] The original sound DSP stores the extracted independent digital accompaniment sequence into its internal transmission buffer. The system then enters processing sub-step S35, which establishes a cross-chip closed-loop backhaul path and performs multi-source data transmission delay compensation.

[0074] The original sound DSP allocates the second set of general-purpose input / output pins and reuses them as an I2S master control transmit bus to send the independent digital accompaniment sequence in the transmit buffer back to the audio effects processing DSP. To compensate for the latency caused by cross-chip transmission and algorithm processing, the audio effects processing DSP allocates a first-in-first-out latency queue in the main memory. The required compensation depth of the queue is determined by the formula for calculating the total number of latency compensation samples in the system:

[0075] ;

[0076] In the formula, This represents the total number of system delay compensation sample points, and its value is limited to a positive integer. This represents the number of inherent algorithm group delay samples generated by the original sound DSP performing the aforementioned band isolation and center cancellation operations. Combined with the filter tap length, it is usually configured as a fixed constant between 256 and 1024. This indicates that the system audio sampling rate has been kept consistent throughout the preceding steps; The underlying bus transmission time when a digital composite audio stream is sent out is determined by dividing the total number of bits of data in a single burst transmission by the aforementioned bus bit clock frequency. This indicates the time taken for the underlying bus transmission during the return of an independent digital accompaniment sequence.

[0077] After acquiring the returned independent digital accompaniment sequence, the system enters processing sub-step S36, performing weighted mixing and dynamic limiting in acoustic removal mode. The multiply-accumulate unit within the audio effects processing DSP simultaneously extracts the independent digital accompaniment sequence and the aligned vocal data. The acoustic removal digital mixing calculation formula within the audio effects processing DSP is as follows:

[0078] ;

[0079] In the formula, Indicates the index number of the current sampling point in the discrete-time series; This represents the current sample value of the final digital mix sequence generated by weighted superposition in the original sound removal mode; This indicates the current sample value of the independent digital accompaniment sequence transmitted back to the master control unit; This represents the current sample value of the digital human voice sequence after processing and performing delayed alignment operations using a first-in-first-out queue. The recombination scaling weight coefficient of the independent accompaniment channel is configured in this embodiment to range from 0.6 to 0.9. The configuration is based on balancing the fullness of the background music and the auditory prominence of the singer's dry voice. This represents the reconstruction scaling weighting coefficient for the delayed vocal channel, with a value range of 0.8 to 1.0. A hardware-level dynamic range compressor is connected in series at the output of the multiply-accumulate unit. When a sample exceeding the safety threshold (set to -3dBFS at full scale) is detected, the output gain is reduced, generating the final digital mix sequence and pushing it to the data conversion unit.

[0080] The system then proceeds to step S40, which involves performing digital-to-analog conversion and output. This specifically includes the following sub-steps:

[0081] S41 performs digital-to-analog domain voltage mapping and filtering reconstruction. The digital-to-analog converter (DAC) integrated within the audio processing DSP receives the final digital mix sequence generated in the preceding steps. The DAC relies on an internal resistor network or switched capacitor array to convert discrete digital quantization values ​​into corresponding stepped voltage distributions. The voltage mapping calculation formula within the DAC is as follows:

[0082] ;

[0083] In the formula, Indicates the index number of the current sampling point in the discrete-time series; This represents the discrete step voltage sample values ​​generated by the mapping; This represents the input digital mix sample value, referring to the final digital mix sequence output from the preceding steps; This indicates the quantization bit width, set to 16 or 24. The denominator is in an absolutely non-zero state to prevent abnormal dead zones in division calculations. This indicates the analog reference voltage configured inside the digital-to-analog converter (DAC). A fixed-level output is provided by an independent low-dropout linear regulator, with a value ranging from 1.8V to 3.3V, specifically determined by the internal power supply rail level of the chip. To filter out high-frequency image aliasing components, an analog low-pass reconstruction filter is connected in series at the back end of the DAC to generate a smooth, continuous-time analog audio signal.

[0084] S42 executes the internal routing transmission and interface output of the analog audio signal. As a preferred method, the continuous-time analog audio signal is transmitted via internal traces on the printed circuit board to the 3.5mm analog interface 5 located on the receiver housing 1. A DC-blocking coupling capacitor is placed between the signal trace and the 3.5mm analog interface 5 on the printed circuit board. Based on the theory of a first-order high-pass filter and considering the input impedance of the external receiving device (approximately 10kΩ), the capacitance of the DC-blocking coupling capacitor is configured in the range of 10μF to 220μF to ensure that the lower frequency band of the audio signal does not experience excessive attenuation. The AC-coupled continuous-time analog audio signal arrives at the 3.5mm analog interface 5, awaiting external transmission.

[0085] S43 is a unidirectional cable routing deployment based on mobile terminal connection scenarios. When the external sound output requirement matches application scenario one, the sound source device is a mobile terminal. The user uses a 3.5mm analog audio cable, with one end inserted into the 3.5mm analog interface 5 on the receiver housing 1, and the other end connected to the analog input terminal of an independent external speaker. Continuous analog audio electrical signals flow unidirectionally into the independent external speaker for sound output.

[0086] S44 is a multi-level bypass routing deployment based on a fixed host connection scenario. When the external sound output requirement matches application scenario two, the audio source device is a smart TV or in-vehicle entertainment system with a built-in speaker array. The user uses a 3.5mm analog audio cable, connecting one end to the receiver's 3.5mm analog interface 5, and the other end directly into the auxiliary analog input interface of the smart TV or in-vehicle entertainment system. The continuous-time analog audio signal, mixed with accompaniment and vocals, is transmitted back to the original audio source device for playback. Optionally, if the device configuration requirements change, the cable can be bypassed to a separate external speaker. After the above deployment is executed, the system completes the distribution operation from digital audio to the physical output end.

[0087] I. Specific Application Example: Karaoke Scenario with External Connection to Home Smart TV

[0088] In the home entertainment scenario, the smart TV serves as the core display and sound source (i.e., the fixed host connection scenario in application scenario two).

[0089] 1. Hardware Connection and Initialization Configuration: The operator presses and holds the power button 4 on the receiver with acoustic cancellation function for more than the set first time threshold (e.g., 3 seconds). Power button 4 generates dual control signals, the DC-DC step-down power module starts supplying power, the audio processing DSP executes the power-on reset sequence, and the system completes initialization. The user uses a double-ended 3.5mm analog cable, inserting one end into the receiver's 3.5mm analog interface 5 and the other end into the analog input interface on the back of the smart TV. Simultaneously, the smart TV establishes a Bluetooth audio transmission link with the receiver through its built-in Bluetooth module, forming a digital composite audio stream input channel.

[0090] 2. Audio Transmission and Signal Processing: The user plays a standard-format online music video (containing the original vocals and background accompaniment in a digital composite audio stream) on a smart TV. At this time, if two users begin singing with wireless microphones:

[0091] To activate the original sound cancellation mode: The user briefly presses the power button 4 on the receiver. The microcontroller internally calculates the actual press pulse duration based on a formula, as follows:

[0092] ;

[0093] When the calculation yields When the signal is between the mechanical jitter filtering threshold (50 milliseconds) and the short press judgment upper limit threshold (1.5 seconds), the microcontroller confirms that the instruction is compliant, sets the routing status flag in the internal non-volatile memory to a logic high level, and triggers the original sound cancellation mode.

[0094] Dual-core collaborative processing: The audio processing DSP cuts off the direct data link, transmitting the received digital composite audio stream from Bluetooth to the acoustic removal DSP via the I2S transmit pin. After receiving the data, the acoustic removal DSP performs acoustic calculations based on the accompaniment extraction calculation formula, as follows:

[0095] ;

[0096] The original sound DSP utilizes the central image cancellation term. Filter out the center sound image component of human voice in the 200Hz-4000Hz frequency band, and rely on the low-frequency preservation coefficient. With high frequency retention factor Complete extraction of independent digital accompaniment sequences Then it is sent back to the audio processing DSP.

[0097] Synchronization and Mixing: The microphone receiving module captures the user's RF vocal signal, down-converts and demodulates it, and then sends it to the audio processing DSP. The audio processing DSP calls the vocal signal enhancement formula, combines the dynamic noise reduction depth coefficient and acoustic attenuation coefficient to perform noise reduction and spatial reverberation processing, generating a processed digital vocal sequence. To address the latency differences caused by cross-chip transmission, the audio processing DSP determines the compensation depth of the first-in-first-out delay queue based on the system's total delay compensation sample number calculation formula, as follows:

[0098] ;

[0099] After queue alignment, the multiply-accumulate unit inside the audio processing DSP calculates the following based on the acoustic removal digital mixing formula:

[0100] ;

[0101] The delayed-aligned vocal sequence and the independent accompaniment sequence are added together according to the set recombination scaling weight coefficient to complete the digital mixing.

[0102] 3. Output playback:

[0103] The DAC unit inside the audio processing DSP receives the final digital mix sequence and calculates it according to the following voltage mapping formula:

[0104] ;

[0105] Discrete digital quantization values ​​are converted into corresponding stepped voltage distributions, and then passed through an analog low-pass reconstruction filter to generate a continuous-time analog audio signal. This signal is sent back to the analog input interface of the smart TV via a 3.5mm analog interface 5 and a cable, where it is played in real time by the built-in speaker, producing a synthesized audio mix of independent accompaniment and the user's vocals.

[0106] II. Experimental Verification and Effect Comparison

[0107] To verify the technical advantages of the dual-processor isolated processing architecture of this invention over the traditional single-processor solution, an Audio Precision APx555 B-series audio analyzer was used for actual testing.

[0108] 1. Test environment and control group settings

[0109] Experimental group: Prototype using the dual-processor architecture (audio processing DSP + original sound removal DSP) of this invention.

[0110] Control group: A prototype with a traditional architecture using a single processor of the same clock frequency (responsible for all sound effects, noise reduction and mixing tasks).

[0111] Test signal: standard 1kHz test sine wave, and standard test audio (WAV format, 48kHz / 24bit) containing 0dBFS wideband accompaniment and -6dBFS centered vocals.

[0112] 2. Test Data Results

[0113] Test Project control group experimental group Test standards and instructions Attenuation of original sound -18.5dB -34.2dB The calculation focuses on extracting residual energy from the centered vocal frequency band. A smaller value indicates more thorough removal of the original sound. End-to-end audio latency 35ms 12ms The time difference between microphone input and DAC analog output is eliminated by using two DSPs in parallel processing in this solution. System signal-to-noise ratio 88dB 102dB The noise reduction algorithm runs independently on the original sound reduction DSP, avoiding the superposition of crosstalk and truncation errors within the digital bus. Total Harmonic Distortion plus Noise 0.85% 0.06% This solution frees up the computing power of the audio processing DSP, enabling it to run higher-order reverb and anti-aliasing filtering algorithms.

[0114] 3. Frequency Domain Effect Simulation and Graph Analysis

[0115] like Figure 7 As shown, the horizontal axis represents frequency (Hz) on a logarithmic scale, and the vertical axis represents power spectral density (dB / Hz). The thin dashed line represents the digital composite audio stream before processing, and the thick solid line represents the independent digital accompaniment sequence after processing. The figure shows that within the vocal dominant frequency band of 300Hz to 3000Hz, the system strictly implements the center image cancellation term. The calculation shows that the energy peaks in this range (thick solid line) are suppressed, with an amplitude attenuation exceeding 30dB, directly reflecting the filtering out of vocal frequency components. Simultaneously, based on the accompaniment extraction calculation formula... and The path, with its thick solid line and thin dashed line, almost completely overlaps in the low-frequency range below 300Hz (such as the bass drum band) and the high-frequency range above 3000Hz (such as the overtone band). This indicates that the solution deeply cancels out vocals while preserving the frequency domain energy distribution at both ends of the original accompaniment without loss.

[0116] 4. Results and Conclusions

[0117] Based on the objective experimental data and frequency domain simulation analysis based on the computational model, the dual-processor physical isolation and multi-source data bus backhaul compensation mechanism constructed in this invention is feasible in terms of technical links. While ensuring the original sound attenuation depth, it compensates for system delay differences through underlying algorithms and completely preserves the frequency band details of the background accompaniment, thus solving the technical problems of incomplete original sound filtering and accompaniment distortion in existing equipment.

Claims

1. A wireless microphone receiver with acoustic cancellation function, comprising a housing (1), an external interaction component disposed in the housing (1), and a circuit system within the housing (1), wherein the external interaction component includes a USB charging port (2), a 3.5mm analog interface (5), an LED indicator (3), and a power button (4), characterized in that: The circuit system includes a power supply and control module. The charging circuit in the power supply and control module is connected to the USB charging interface (2) and the lithium battery assembly. The lithium battery assembly is connected to the DC-DC step-down power supply module, which converts the output voltage of the lithium battery assembly into the 3.3V DC operating voltage required by the subsequent chip. The power button (4) is synchronously connected to the enable terminal of the DC-DC step-down power supply module and the general input pin of the audio processing DSP in the core processing cluster of the circuit system. The audio processing DSP is connected to the LED indicator (3), the 3.5mm analog interface (5), and the Bluetooth front-end unit in the wireless communication and receiving module included in the circuit system. The microphone receiving module in the wireless communication and receiving module is connected to the audio processing DSP via an I2S bus. The audio processing DSP and the original sound removal DSP included in the core processing cluster establish a cross-chip data link through a bidirectional I2S audio bus and a UART communication bus.

2. A wireless microphone receiver with acoustic cancellation function according to claim 1, characterized in that, When the duration of pressing the power button (4) exceeds the set first time threshold, the power button (4) closes the underlying circuit and generates dual control signals simultaneously. One signal branch is transmitted to the enable pin of the DC-DC step-down power supply module, and the other signal branch is transmitted to the general-purpose input pin of the audio processing DSP. After the microcontroller unit inside the audio processing DSP is connected to the working voltage, it executes a power-on reset sequence, loads the boot firmware from the internal non-volatile memory, sets the phase-locked loop clock tree of the system, and initializes the communication pins of each data bus to a high-impedance state. The audio processing DSP controls the LED indicator (3) to output a high level through its driving pin, thereby driving the LED indicator (3) to be in a continuously conducting and luminous state. The audio processing DSP uses its internally integrated analog-to-digital converter to collect the current end-face output voltage of the lithium battery component. The first time threshold is a preset time constant in the internal non-volatile memory, which is used to represent the duration required for the power button (4) to close the underlying circuit.

3. A wireless microphone receiver with acoustic cancellation function according to claim 2, characterized in that, The audio processing DSP sends address bytes to the connected microphone receiving module via the I2C communication bus and writes multiple sets of operating parameters to the control register; The written multiple sets of operating parameters specify the sampling rate and bit depth of the I2S bus between the microphone receiving module and the audio processing DSP, and set the initial wireless frequency band for the operation of the RF front-end circuit. The audio processing DSP sends an initialization command in data frame format to the original sound removal DSP via the UART communication bus. After receiving the initialization command, the original sound removal DSP clears its internal audio data buffer and resets the bidirectional I2S audio bus transmit / receive state that is closed-loop connected to the audio processing DSP.

4. A wireless microphone receiver with acoustic cancellation function according to claim 1, characterized in that, The Bluetooth front-end unit establishes a wireless connection with an external audio source device and receives a digital composite audio stream containing human voice and accompaniment information. The audio processing DSP opens a circular buffer in its internal random access memory and writes the digital composite audio stream into the circular buffer. The microphone receiving module drives the external radio frequency antenna to scan the spatial channel, receive the radio frequency human voice signal emitted by the independent wireless microphone, and down-convert the received radio frequency human voice signal to the baseband. The analog-to-digital converter unit inside the audio processing DSP resamples the baseband signal and demodulates it to produce a digital human voice sequence containing the user's voice characteristics.

5. A wireless microphone receiver with acoustic cancellation function according to claim 4, characterized in that, The audio processing DSP calls the internally embedded digital signal processing algorithm to perform noise reduction, equalization and spatial reverberation operations on the digital human voice sequence to generate the processed digital human voice sequence. The audio processing DSP uses an internal hardware timer to compare the generation timestamp of the processed digital human voice sequence with the arrival timestamp of the digital composite audio stream in the circular buffer, and performs synchronization alignment operations such as silencing, zero padding, or discarding old data frames.

6. A wireless microphone receiver with acoustic cancellation function according to claim 5, characterized in that, The microcontroller inside the audio processing DSP continuously monitors the level status of the general-purpose input pin connected to the power button (4), and the microcontroller performs a toggle operation on the routing status flag bit set in the internal non-volatile memory. When the routing status flag is at a logic low level, it is determined that the original sound mode is in place, and the microcontroller controls the internal audio bus to close the direct connection between the external Bluetooth channel and the internal mixing calculation unit. The digital composite audio stream, which is stored in the circular buffer and has completed synchronization alignment, is directly fed into the multiply-accumulate unit inside the audio processing DSP.

7. A wireless microphone receiver with acoustic cancellation function according to claim 5, characterized in that, When the routing status flag is at a logic high level, it is determined that the original sound cancellation mode is entered. The audio effect processing DSP changes the internal audio bus topology and disconnects the data direct link between the ring buffer and the internal multiply-accumulate operation unit. The audio processing DSP sends the digital composite audio stream to the original sound removal DSP through the bidirectional I2S audio bus. The data parsing module inside the original sound removal DSP decomposes the multiplexed data frame into independent left channel audio sequences and right channel audio sequences. The digital crossover frequency divider integrated inside the original sound removal DSP divides the separated channel data into three independent frequency band branches: low frequency, mid frequency and high frequency. The original sound DSP performs energy normalization on the left and right channel sequences in the mid-frequency band, filters out the vocal frequency band components to generate an independent digital accompaniment sequence, and stores it in the internal transmission buffer.

8. A wireless microphone receiver with acoustic cancellation function according to claim 7, characterized in that, The original sound removal DSP allocates the second set of general-purpose input and output pins and reuses them as an I2S master control transmission bus to send the independent digital accompaniment sequence in the transmission buffer back to the sound effect processing DSP. The audio processing DSP allocates a first-in-first-out delay queue in the main memory. The multiply-accumulate unit inside the audio processing DSP synchronously extracts the independent digital accompaniment sequence and the processed digital vocal sequence after delay alignment operation by the first-in-first-out delay queue, and adds them together to generate the final digital mix sequence.

9. A wireless microphone receiver with acoustic cancellation function according to claim 8, characterized in that, The DAC (Digital-to-Analog Converter) unit integrated within the audio processing DSP receives the final digital mixing sequence and maps the final digital mixing sequence into a continuous-time analog audio electrical signal. The DAC digital-to-analog conversion unit relies on an internal resistor network or switched capacitor array to convert discrete digital quantization values ​​into corresponding stepped voltage distributions. The DAC digital-to-analog conversion unit is connected in series with an analog low-pass reconstruction filter at the back end to generate a continuous-time analog audio electrical signal. The continuous-time analog audio signal is transmitted to the 3.5mm analog interface (5) on the receiver housing end face via the internal traces of the printed circuit board. The printed circuit board is provided with a DC blocking coupling capacitor between the signal traces and the 3.5mm analog interface (5).

10. A wireless microphone receiver with acoustic cancellation function according to claim 9, characterized in that, The 3.5mm analog interface (5) is used to adapt to an external 3.5mm analog audio cable. The 3.5mm analog interface (5) is connected to an independent external speaker through the 3.5mm analog audio cable, so that the continuous time analog audio electrical signal flows unidirectionally into the independent external speaker to produce sound. Alternatively, the 3.5mm analog interface (5) can be connected to the auxiliary analog input interface of a smart TV or in-vehicle entertainment host via the 3.5mm analog audio cable to transmit the continuous-time analog audio signal back to the original audio source device for playback.

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