Method and system for converting low sampling rate of audio data into high sampling rate and medium

Abstract

The invention discloses a method and a system for converting audio data from a low sampling rate to a high sampling rate, and a medium. The method comprises the following steps: acquiring a digital audio signal with a low sampling rate; counting the current sampling frequency of the digital audio signal; according to the current sampling frequency and the target interpolation frequency, various interpolation parameters required by odd-even statistical fractional interpolation calculation are searched from a parameter lookup table; performing oversampling processing on the obtained digital audio signal to obtain an oversampling data sequence, and determining an interpolation coefficient according to each interpolation parameter; and reading corresponding data from the oversampling data sequence according to an odd-even crossing sequence and the interpolation coefficient, and forming a new data frame to obtain a data sequence of the digital audio signal with the high sampling rate. According to the invention, audios from different sampling rates received by the multifunctional card are unified to be output at the same rate, lossless output of converted audio signals is ensured, any frequency input can be compatible, and only a small number of logic units and storage space are used while high performance is considered.

Description

Technical Field

[0001] This invention relates to the field of audio data processing technology, and in particular to a method, system, and medium for converting low-sampling-rate audio data to high-sampling-rate data. Background Technology

[0002] Sound waves are continuous waveform signals, but in the process of digitization (i.e. A / D conversion), it is impossible to store the entire sound signal. Instead, the waveform is sampled and quantized at intervals according to the set sampling rate and quantization precision, and the quantized value of this set of sampled data is obtained. Therefore, different audio sampling rates (sampling frequency) and quantization bit widths are generated. Common sampling rates are shown in Table 1.

[0003] In traditional multifunction card designs, the audio output interface hardware can only play audio data at a fixed frequency, such as 96kHz or 48kHz. However, in application scenarios, the multifunction card needs to receive audio data acquired from different transmitting cards. The audio frequencies acquired by the transmitting cards may be 32kHz, 44.1kHz, 48kHz, etc. If the bitrates do not match, it will cause noise, popping, audio distortion, and other audio discrepancies during playback, making it impossible to accurately play the acquired audio information. To ensure that the playback audio frequency of the multifunction card matches the audio frequency acquired by the transmitting card, different firmware specifications for the multifunction card's playback frequencies need to be designed, which greatly increases the development and equipment costs and reduces the device compatibility of the multifunction card.

[0004] Table 1. Application Scenarios and Characteristics of Audio Sampling Rate

[0005]

[0006] Summary of the Invention

[0007] The main objective of this invention is to propose an OESFI (Odd-Even Statistical Fractional Interpolation) interpolation method for rate adaptation of low-sampling-rate audio digital signals to high-sampling-rate audio digital signals based on a multifunction card. Specifically, it involves a method, system, and medium for converting low-sampling-rate audio data to high-sampling-rate data. The aim is to achieve audio rate adaptation, unify audio received by the multifunction card from different sampling rates to the same output rate, and ensure lossless output of the converted audio signal. It is compatible with arbitrary frequency input, and while achieving high performance, it uses only a small number of logic units and storage space, thereby enhancing product competitiveness.

[0008] To achieve the above objectives, the present invention provides a method for converting low-sampling-rate audio data to high-sampling-rate audio data, the method comprising the following steps:

[0009] Step S10: Acquire a low-sampling-rate digital audio signal;

[0010] Step S20: Calculate the current sampling frequency f of the digital audio signal. s ;

[0011] Step S30, according to the current sampling frequency f s and target interpolation frequency f t The interpolation parameters required for the parity-even statistical fraction interpolation method are retrieved from the parameter lookup table. These interpolation parameters include the target interpolation number p, the channel rate level K, and the step size H. i The total number of step size values ​​S i And the number n of various step sizes;

[0012] Step S40: Perform m-fold oversampling on the acquired digital audio signal to obtain an oversampled data sequence, and determine the interpolation coefficients according to the interpolation parameters.

[0013] Step S50: According to the odd-even crossover order, the corresponding data is read from the oversampled data sequence based on the interpolation coefficients and a new data frame is formed to obtain the data sequence of the high sampling rate digital audio signal.

[0014] A further technical solution of the present invention is that the calculation formula used in step S40, which determines the interpolation coefficients based on the interpolation parameters, is as follows:

[0015]

[0016] Where p is the target interpolation number, K is the channel rate level, and H... i S is the step size value. i The total number of values ​​for that step size is n, the number of various step sizes is m, and the oversampling factor is m.

[0017] A further technical solution of the present invention is that step S50 includes:

[0018] Step S501: Based on the interpolation coefficients, select the step size values ​​H1…H using alternating odd and even or cyclic order. n The oversampled data sequence is address-stepped and data points are extracted according to the address to generate a data sequence of a high-sampling-rate digital audio signal.

[0019] A further technical solution of the present invention is that step S501 includes:

[0020] Step S502: Initialize an address pointer;

[0021] Step S503: Select the step size value H according to the alternation of odd and even or the cyclical order. i ;

[0022] Step S504: For each selected step size value, the address pointer is incremented by the step size value, and the data point at the position pointed to by the address pointer is extracted from the oversampled data sequence. At the same time, the used count counter corresponding to the step size value is incremented by one.

[0023] Repeat steps S502 to S504 until all step size values ​​have been used a total of S times for that step size value. i Or the total number of data points extracted reaches the target interpolation number p.

[0024] A further technical solution of the present invention is that step S20 includes:

[0025] The current sampling frequency f of the digital audio signal is inferred based on the number of Ethernet packets received per unit time or the total number of bytes in the Ethernet packets. s .

[0026] A further technical solution of the present invention is that, after step S10, it further includes:

[0027] Step S101: Determine whether the digital audio signal is an audio data packet based on the information field of the Ethernet packet header;

[0028] Step S102: If it is an audio data packet, then buffer the digital audio signal;

[0029] In step S103, if the packet is not an audio data packet, the Ethernet packet is discarded.

[0030] A further technical solution of the present invention is that, after step S50, it further includes:

[0031] Step S60: Store the data sequence of the high-sampling-rate digital audio signal into a first-in-first-out (FIFO) queue for reading and output.

[0032] A further technical solution of the present invention is that, after step S60, it further includes:

[0033] According to the output timing requirements of the audio output interface, audio data is retrieved from the first-in-first-out (FIFO) queue and output.

[0034] To achieve the above objectives, the present invention also proposes a system for converting low-sampling-rate audio data to high-sampling-rate audio data. The system includes a memory, a processor, and a program for converting low-sampling-rate audio data to high-sampling-rate audio data stored on the processor. The program for converting low-sampling-rate audio data to high-sampling-rate audio data is executed by the processor to perform the steps of the method described above.

[0035] To achieve the above objectives, the present invention also proposes a computer-readable storage medium storing a program for converting audio data from a low sampling rate to a high sampling rate, wherein the program is executed by a processor to perform the steps of the method described above.

[0036] The beneficial effects of the method, system, and medium for converting low-sampling-rate audio data to high-sampling-rate in this invention are:

[0037] 1. This invention can be adapted to convert any low sampling rate data input to high sampling rate data output of a multifunction card, thus enhancing device compatibility;

[0038] 2. The audio interpolation method based on statistical averaging of odd and even interpolation used in this invention is easy to implement logically;

[0039] 3. This invention has a concise logic, occupies only a small amount of logic resources, and is easy to deploy;

[0040] 4. This invention has high conversion accuracy and good playback effect after conversion. Attached Figure Description

[0041] 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 the structures shown in these drawings without creative effort.

[0042] Figure 1 This is a system block diagram of the method for converting low-sampling-rate audio data to high-sampling-rate according to the present invention;

[0043] Figure 2 This is a flowchart illustrating the first embodiment of the method for converting low-sampling-rate audio data to high-sampling-rate according to the present invention;

[0044] Figure 3 This is a detailed flowchart of step S501;

[0045] Figure 4 This is a timing diagram demonstrating the interpolation process.

[0046] Figure 5 This is a demonstration of the interpolation process - model diagram;

[0047] Figure 6 This is a flowchart illustrating the second embodiment of the method for converting low-sampling-rate audio data to high-sampling-rate according to the present invention.

[0048] The realization of the objective, functional features and advantages of the present invention will be further explained in conjunction with the embodiments and with reference to the accompanying drawings. Detailed Implementation

[0049] 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 a part of the embodiments of the present invention, and not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those of ordinary skill in the art without creative effort are within the scope of protection of the present invention.

[0050] To unify the output of audio from different sampling rates received by a multifunction card to the same rate while ensuring the quality of the converted audio, improving the compatibility of the multifunction card device, and reducing audio adaptation costs, this invention proposes a method for converting low-sampling-rate audio data to high-sampling-rate audio. The technical solution mainly employs the OESFI audio interpolation method implemented on the multifunction card's FPGA, using software to complete audio rate adaptation and ensure lossless audio signal output. It is compatible with arbitrary frequency input, achieving high performance while using only a small number of logic units and storage space, thus enhancing product competitiveness.

[0051] Specifically, this invention proposes a method for converting low-sampling-rate audio data to high-sampling-rate audio data. The following first introduces the system on which this method for converting low-sampling-rate audio data to high-sampling-rate audio data operates.

[0052] like Figure 1 As shown, the system operating under the method for converting low-sampling-rate audio data to high-sampling-rate according to the present invention includes an audio interpolation processing module and a subsequent audio interface module. The audio interpolation processing module includes the following sub-modules: an audio packet filtering module, an audio frequency statistics module, an audio packet cache RAM module, an audio lookup table module, an OESFI interpolation processor, an OESFI cache FIFO module, and an audio interface module.

[0053] The overall process of the system running the low sampling rate to high sampling rate audio data conversion method of the present invention is as follows: The audio interpolation processing module first receives the digital audio signal collected from the sending card through the network port or other interface. The received data is first filtered by the audio packet filtering module to extract the audio data packets, and then enters the audio packet buffer RAM module and the audio frequency statistics module respectively. After the audio frequency statistics module calculates the audio frequency, it looks up the various parameters calculated by OESFI from the parameter lookup table according to the current sampling frequency and the target interpolation frequency. Then, the OESFI interpolation processor determines the interpolation coefficients according to the parameters in the lookup table, retrieves the audio data from the audio packet buffer RAM module, and stores the interpolated data into the OESFI buffer FIFO module (first in first out queue) for the subsequent audio interface module to read and output.

[0054] The audio interface module retrieves audio data from the FIFO of the audio interpolation module and outputs the audio data according to the output timing requirements of the audio output interface (such as the IIS interface). Then, the audio output interface can be connected through audio equipment such as headphones and speakers to hear the audio information collected by the sending card.

[0055] The functions of each submodule of the audio interpolation processing module are as follows:

[0056] Audio packet filtering module: This module receives data through the network port and determines whether it is an audio packet based on the information fields in the Ethernet packet header. If it is, it stores the data in the audio packet buffer RAM module and simultaneously sends the data to the audio frequency statistics module; otherwise, it discards the Ethernet packet.

[0057] Audio Frequency Statistics Module: Since the sending mechanism of the Ethernet packets at the transmitting end involves collecting 768 bytes of audio data and then sending that 768 bytes to the receiving end via the network port, and because the sampling frequency is fixed, the Ethernet packet transmission time interval is also fixed, and the data size of each packet is also a fixed 768 bytes. The audio frequency statistics module can deduce the sampling frequency of the transmitting end by counting the number of Ethernet packets received within 1 second or the number of bytes received within 100ms. Table 2 shows the packet transmission statistics for the transmitting segment under some sampling frequencies.

[0058] Table 2 shows the statistics of message data at some sampling rates.

[0059]

[0060] Audio Packet Cache RAM Module: This module instantiates an 8-bit*1024-depth RAM (RandomAccess Memory) storage device to temporarily store 768 bytes of audio data received from the Ethernet interface.

[0061] Audio Lookup Table Module: This module internally instantiates a ROM (Read-Only Memory) to store the parameter information of the lookup table. Different address addresses are used to look up different sampling rate conversion parameters. For example, addresses 0x0000 to 0x003F store the target interpolation number, step size, and total number of step sizes for a target sampling rate of 97.656kHz. Taking two step sizes for interpolation and a target conversion sampling rate of 97.656kHz as an example, the parameters for some sampling rates are shown in Table 3 below.

[0062] Table 3 shows the parameters for converting some sampling rates to 97.656kHz.

[0063]

[0064] OESFI Interpolation Processor: This module uses the frequency statistics module to count the transmitting end sampling frequency and the target interpolation frequency, retrieves the corresponding parameters from the lookup table, and then, based on the retrieved parameters such as the step size and the total number of step sizes, retrieves data from the audio packet buffer RAM module and stores it in the OESFI buffer FIFO. The specific implementation process is described later.

[0065] OESFI cache FIFO module: This module instantiates a FIFO (First In First Out) memory to temporarily store interpolated data for use by subsequent modules.

[0066] Audio Interface Module: This module is a timing conversion module. Based on the output frequency, it reads data from the OESFI buffer FIFO and outputs it to the broadcasting device according to the IIS timing.

[0067] like Figure 2 As shown, the first embodiment of the method for converting low-sampling-rate audio data to high-sampling-rate according to the present invention includes the following steps:

[0068] Step S10: Acquire a low-sampling-rate digital audio signal.

[0069] In this embodiment, a low-sampling-rate digital audio signal is obtained through the network port of the audio packet receiving module.

[0070] Step S20: Calculate the current sampling frequency f of the digital audio signal. s .

[0071] In this embodiment, after obtaining a low-sampling-rate digital audio signal through the network port of the audio packet receiving module, the current sampling frequency f of the digital audio signal is then calculated by the audio frequency statistics module. s .

[0072] Step S20 specifically includes:

[0073] The current sampling frequency f of the digital audio signal is inferred based on the number of Ethernet packets received per unit time or the total number of bytes in the Ethernet packets. s .

[0074] For example, the audio frequency statistics module can infer the sampling frequency of the transmitting end by counting the number of Ethernet packets received within 1 second or the number of bytes received within 100ms.

[0075] Step S30, according to the current sampling frequency f s and target interpolation frequency f t The interpolation parameters required for the parity-even statistical fraction interpolation method are retrieved from the parameter lookup table. These interpolation parameters include the target interpolation number p, the channel rate level K, and the step size H. i The total number of step size values ​​S i And the number of various step sizes, n.

[0076] Step S40: Perform m-fold oversampling on the acquired digital audio signal to obtain an oversampled data sequence, and determine the interpolation coefficients based on the interpolation parameters.

[0077] The calculation formula used in step S40, which determines the interpolation coefficients based on the interpolation parameters, is as follows:

[0078]

[0079] Where p is the target interpolation number, K is the channel rate level, and H... i S is the step size value. i The total number of values ​​for that step size is n, the number of various step sizes is m, and the oversampling factor is m.

[0080] Step S50: According to the odd-even crossover order, the corresponding data is read from the oversampled data sequence based on the interpolation coefficients and a new data frame is formed to obtain the data sequence of the high sampling rate digital audio signal.

[0081] Step S50 specifically includes:

[0082] Step S501: Based on the interpolation coefficients, select the step size values ​​H1…H using alternating odd and even or cyclic order. n The oversampled data sequence is address-stepped and data points are extracted according to the address to generate a data sequence of a high-sampling-rate digital audio signal.

[0083] like Figure 3 As shown, step S501 specifically includes:

[0084] Step S502: Initialize an address pointer;

[0085] Step S503: Select the step size value H according to the alternation of odd and even or the cyclical order. i ;

[0086] Step S504: For each selected step size value, the address pointer is incremented by the step size value, and the data point at the position pointed to by the address pointer is extracted from the oversampled data sequence. At the same time, the used count counter corresponding to the step size value is incremented by one.

[0087] Step S505: Repeat steps S502 to S504 until all step size values ​​have been used a total of S times for that step size value. i Or the total number of data points extracted reaches the target interpolation number p.

[0088] The specific OESFI interpolation process involved in this embodiment is as follows:

[0089] The OESFI interpolation coefficient calculation formula is shown in equation (1). The system first performs m times oversampling on the input audio data, where p is the target interpolation number, K is the channel rate level (i.e., the amount of channel data contained in each Ethernet packet), and H... i and S i These represent the step size and the total number of steps, respectively. The step size is determined by an address pointer pointing to the oversampled data storage space after oversampling. The increment of the address pointer after each value is the step size. t and f s These represent the target interpolation frequency and the current sampling frequency, respectively, where n is the number of various step sizes. The above coefficients can be obtained by looking up the statistical values ​​from the frequency statistics module.

[0090] To simplify the calculation, we take two step sizes as examples (n=2), K=128, target sampling rate is 97.656kHz, current sampling rate is 48kHz, as shown in equation (2). Calculation yields p=261 (rounded up), oversampling rate is 8 (i.e., the original audio data is copied 8 times, A0-->A0~A7), after oversampling, the original audio data becomes K*8=128*8=1024 channel data, and p=261 data need to be extracted from these 1024 data. It is required to distribute them as evenly as possible among the 1024 data. Then, according to the fraction ratio, the average is 1024 / 261=3.9234. Taking the average step size of 3.9234 and rounding up and down, we get H1=4, H2=3. According to formula 3-1, the proportion of H1 is S1=241, and the proportion of H2 is S2=20. Figure 4As shown, let the initial address be 0. Take out the data A0 corresponding to address 0 in the 1024 space. After the first increment of H1, we get address 4 and take out the data A4 corresponding to address 4. After the second increment of H2, we get address 7 and take out the data A7 corresponding to address 7. After the third increment of H1, we get address 11 and take out the data B3 corresponding to address 11. We increase the step size H1 and H2 in odd-even rounds. After each increment of step size H1, the counter of H1 is incremented by 1. The same applies to H2. When the counter of H2 counts to S2, the step size of H2 is no longer used. We continue to increase the step size of H1 until the counter of H1 counts to S1 and the current round of interpolation ends.

[0091]

[0092] like Figure 4 and Figure 5 This is a schematic diagram of the OESFI interpolation process. The original data A to E, after being oversampled by 8 times, present 5*8=40 original data points to be interpolated. Under step size control, the interpolation address is incremented sequentially. After simplification to dual-step interpolation, data is retrieved from memory in an alternating odd-even order and assembled into a new data frame. The interpolated frequency is increased to 2.4 times (12 / 5) of the original frequency. The target interpolation frequency is based on a statistical average, and the input frequency is statistically analyzed in real time. The input source can be switched arbitrarily during use.

[0093] based on Figure 2 The first embodiment shown presents a second embodiment of the method for converting low-sampling-rate audio data to high-sampling-rate, as described in the first embodiment. Figure 6 As shown, this embodiment is similar to Figure 2 The difference in the first embodiment shown is that, in this embodiment, the following step is included after step S10:

[0094] Step S101: Determine whether the digital audio signal is an audio data packet based on the information field of the Ethernet packet header.

[0095] Step S102: If it is an audio data message, then the digital audio signal is buffered.

[0096] In step S103, if the packet is not an audio data packet, the Ethernet packet is discarded.

[0097] In this embodiment, the audio packet filtering module receives data through the network port and determines whether it is an audio data packet based on the information field of the Ethernet packet header. If it is, it stores the data in the audio packet cache RAM module and simultaneously sends the data to the audio frequency statistics module; otherwise, it discards the Ethernet packet.

[0098] In this embodiment, the method further includes the following after step S50:

[0099] Step S60: Store the data sequence of the high-sampling-rate digital audio signal into a first-in-first-out (FIFO) queue for reading and output.

[0100] Following step S60, the following is also included:

[0101] Step S70: According to the output timing requirements of the audio output interface, retrieve audio data from the first-in-first-out (FIFO) queue and output it.

[0102] The audio interface module retrieves audio data from the FIFO of the audio interpolation module and outputs the audio data according to the output timing requirements of the audio output interface (such as the IIS interface). Then, the audio output interface can be connected through audio equipment such as headphones and speakers to hear the audio information collected by the sending card.

[0103] The beneficial effects of the method for converting low-sampling-rate audio data to high-sampling-rate in this invention are:

[0104] 1. This invention can be adapted to convert any low sampling rate data input to high sampling rate data output of a multifunction card, thus enhancing device compatibility;

[0105] 2. The audio interpolation method based on statistical averaging of odd and even interpolation used in this invention is easy to implement logically;

[0106] 3. This invention has a concise logic, occupies only a small amount of logic resources, and is easy to deploy;

[0107] 4. This invention has high conversion accuracy and good playback effect after conversion.

[0108] To achieve the above objectives, the present invention also proposes a system for converting low-sampling-rate audio data to high-sampling-rate audio data. The system includes a memory, a processor, and a program for converting low-sampling-rate audio data to high-sampling-rate audio data stored on the processor. When the processor runs the program for converting low-sampling-rate audio data to high-sampling-rate audio data, it executes the steps of the method described above, which will not be repeated here.

[0109] To achieve the above objectives, the present invention also proposes a computer-readable storage medium storing a program for converting audio data from a low sampling rate to a high sampling rate. When the program for converting audio data from a low sampling rate to a high sampling rate is run by a processor, the steps of the method described above are executed, and will not be repeated here.

[0110] The above description is only a preferred embodiment of the present invention and does not limit the patent scope of the present invention. All equivalent structural transformations made under the concept of the present invention using the contents of the present invention specification and drawings, or direct / indirect applications in other related technical fields, are included within the patent protection scope of the present invention.

Claims

1. A method for converting low-sampling-rate audio data to high-sampling-rate audio data, characterized in that, The method includes the following steps: Step S10: Acquire a low-sampling-rate digital audio signal; Step S20: Calculate the current sampling frequency f of the digital audio signal. s ; Step S30, according to the current sampling frequency f s and target interpolation frequency f t The interpolation parameters required for the parity-even statistical fraction interpolation method are retrieved from the parameter lookup table. These interpolation parameters include the target interpolation number p, the channel rate level K, and the step size H. i The total number of step size values ​​S i And the number n of various step sizes; Step S40: Perform m-fold oversampling on the acquired digital audio signal to obtain an oversampled data sequence, and determine the interpolation coefficients according to the interpolation parameters. Step S50: According to the odd-even crossover order, the corresponding data is read from the oversampled data sequence based on the interpolation coefficients and a new data frame is formed to obtain the data sequence of the high sampling rate digital audio signal.

2. The method for converting low-sampling-rate audio data to high-sampling-rate according to claim 1, characterized in that, The calculation formula used in step S40, which determines the interpolation coefficients based on the interpolation parameters, is as follows: Where p is the target interpolation number, K is the channel rate level, and H... i S is the step size value. i The total number of values ​​for that step size is n, the number of various step sizes is m, and the oversampling factor is m.

3. The method for converting low-sampling-rate audio data to high-sampling-rate according to claim 2, characterized in that, Step S50 includes: Step S501: Based on the interpolation coefficients, select the step size values ​​H1…H using alternating odd and even or cyclic order. n The oversampled data sequence is address-stepped and data points are extracted according to the address to generate a data sequence of a high-sampling-rate digital audio signal.

4. The method for converting low-sampling-rate audio data to high-sampling-rate according to claim 3, characterized in that, Step S501 includes: Step S502: Initialize an address pointer; Step S503: Select the step size value H according to the alternation of odd and even or the cyclical order. i ; Step S504: For each selected step size value, the address pointer is incremented by the step size value, and the data point at the position pointed to by the address pointer is extracted from the oversampled data sequence. At the same time, the used count counter corresponding to the step size value is incremented by one. Repeat steps S502 to S504 until all step size values ​​have been used a total of S times for that step size value. i Or the total number of data points extracted reaches the target interpolation number p.

5. The method for converting low-sampling-rate audio data to high-sampling-rate according to claim 1, characterized in that, Step S20 includes: The current sampling frequency f of the digital audio signal is inferred based on the number of Ethernet packets received per unit time or the total number of bytes in the Ethernet packets. s .

6. The method for converting low-sampling-rate audio data to high-sampling-rate according to claim 1, characterized in that, The process following step S10 also includes: Step S101: Determine whether the digital audio signal is an audio data packet based on the information field of the Ethernet packet header; Step S102: If it is an audio data packet, then buffer the digital audio signal; In step S103, if the packet is not an audio data packet, the Ethernet packet is discarded.

7. The method for converting low-sampling-rate audio data to high-sampling-rate according to any one of claims 1 to 6, characterized in that, The step S50 is followed by: Step S60: Store the data sequence of the high-sampling-rate digital audio signal into a first-in-first-out (FIFO) queue for reading and output.

8. The method for converting low-sampling-rate audio data to high-sampling-rate according to claim 7, characterized in that, Following step S60, the following is also included: According to the output timing requirements of the audio output interface, audio data is retrieved from the first-in-first-out (FIFO) queue and output.

9. A system for converting low-sampling-rate audio data to high-sampling-rate audio data, characterized in that, The system includes a memory, a processor, and a program for converting low-sampling-rate audio data to high-sampling-rate audio data stored on the processor, the program being executed by the processor to perform the steps of the method as described in any one of claims 1 to 8.

10. A computer-readable storage medium, characterized in that, The computer-readable storage medium stores a program for converting audio data from a low sampling rate to a high sampling rate, the program being executed by a processor to perform the steps of the method as described in any one of claims 1 to 8.

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