Audio playing method, audio SOC chip and electronic equipment

Abstract

The embodiment of the invention discloses an audio playing method, an audio SOC chip and electronic equipment, and relates to the field of audio processing. The method comprises the following steps: reading a PCM audio digital sequence from a memory; the PCM audio digital sequence comprises a plurality of N-bit sampling points, and N is an integer greater than 1; each N-bit sampling point is converted into a plurality of 1-bit sampling points through a DSM modulator; converting each 1-bit sampling point into a 1-bit analog signal by using a 1-bit analog quantizer; performing amplification processing on the 1-bit analog by using a power amplifier; and playing the amplified 1-bit analog signal through a sounding unit. By replacing a DAC (Digital-to-Analog Converter) in the prior art with the DSM modulator and the 1-bit analog quantizer, the technical effects of reducing the occupied chip area, improving the audio conversion efficiency, optimizing the audio playing quality, simplifying the system design and the like are achieved.

Description

Technical Field

[0001] This application relates to the field of audio processing, and particularly to an audio playback method, an audio SOC chip, and an electronic device. Background Art

[0002] In modern electronic devices, the system-on-chip (SOC) serves as a core component, undertaking multiple tasks such as data processing and signal conversion. Among them, audio signal processing is an important part of the SOC chip's functions, directly related to the quality of the user's auditory experience. To achieve high-quality audio playback, a dedicated digital-to-analog converter (DAC) is usually integrated inside the SOC chip. The role of the DAC is to convert digital audio signals into analog audio signals, which is a crucial step for audio signals to be perceivable by the human auditory system.

[0003] Digital audio signals, as the standard format for modern audio storage and transmission, have advantages such as strong anti-interference ability, easy editing and storage. However, to drive audio output devices such as speakers or headphones, these digital signals must be converted into analog signals. The DAC is precisely the bridge to complete this conversion process, and its performance directly affects the quality of the final audio output, including key indicators such as sound clarity, dynamic range, and signal-to-noise ratio.

[0004] In the design of SOC chips, to ensure that the DAC can achieve the expected audio playback performance, such as the ability to restore high-resolution audio and low distortion, a relatively large chip area often needs to be allocated to it. This is because high-performance DACs require complex circuit structures, including precise current sources, weight amplifiers, switch arrays, and output buffers. The precise layout and mutual matching of these components are crucial for reducing noise, improving linearity, and conversion accuracy. Therefore, although the DAC is indispensable for improving audio quality, its occupation of the SOC chip area has also become an important factor that designers need to weigh. Summary of the Invention

[0005] Embodiments of this application provide an audio playback method, an audio SOC chip, and an electronic device, which can solve the problem that the DAC occupies a large area in the chip in the prior art. The technical solutions are as follows:

[0006] In a first aspect, embodiments of this application provide an audio playback method, which includes:

[0007] Reading a PCM audio digital sequence from a memory; the PCM audio digital sequence includes multiple N-bit sampling points, and N is an integer greater than 1;

[0008] Each N-bit sample point is converted into multiple 1-bit sample points through a DSM modulator;

[0009] Each 1-bit sample point is converted into a 1-bit analog signal by a 1-bit analog quantizer;

[0010] The 1-bit analog is amplified by a power amplifier;

[0011] The amplified 1-bit analog signal is played through a sound generating unit.

[0012] In a second aspect, an audio SOC chip provided by an embodiment of the present application includes:

[0013] A DSM modulator for reading a PCM audio digital sequence from a memory; the PCM audio digital sequence includes multiple N-bit sample points, where N is an integer greater than 1; each N-bit sample point is converted into multiple 1-bit sample points;

[0014] A 1-bit analog quantizer for converting each 1-bit sample point into a 1-bit analog signal;

[0015] A power amplifier for amplifying the 1-bit analog and playing the amplified 1-bit analog signal through a sound generating unit.

[0016] In a fourth aspect, an electronic device provided by an embodiment of the present application may be a device with audio processing functions such as a tablet computer, a personal computer, a mobile phone, etc. It may include: a memory, the above-mentioned audio SOC chip, and a sound generating unit.

[0017] The beneficial effects brought by the technical solutions provided by some embodiments of the present application at least include:

[0018] Since high-performance DACs require complex circuit structures, including precise current sources, weight amplifiers, switch arrays, and output buffers, etc., these components occupy a relatively large chip area. In contrast, the DSM modulator and 1-bit analog quantizer in this solution have relatively simple structures, which can effectively reduce the chip area occupation, thereby reducing costs and improving production efficiency. The DSM modulator converts each N-bit sampling point into multiple 1-bit sampling points, utilizing oversampling and noise shaping techniques, and can achieve high audio quality at a relatively low sampling rate. At the same time, the 1-bit analog quantizer directly converts each 1-bit sampling point into an analog signal, reducing information loss during the conversion process and improving the efficiency of audio conversion. This solution can also achieve high-quality audio playback by optimizing the circuit structure and signal processing flow. In particular, the noise shaping ability of the DSM modulator can reduce in-band noise and improve the signal-to-noise ratio while ensuring audio quality, thereby enhancing the user's auditory experience. After adopting the DSM modulator and 1-bit analog quantizer, the overall design of the audio system becomes more concise and clear. This not only reduces the complexity of system design but also improves the reliability and stability of the system. Brief Description of the Drawings

[0019] To more clearly illustrate the technical solutions in the embodiments of the present application or the prior art, the following will briefly introduce the drawings required for use in the description of the embodiments or the prior art. Obviously, the drawings in the following description are only some embodiments of the present application. For those of ordinary skill in the art, without creative efforts, other drawings can be obtained based on these drawings.

[0020] Figure 1 is a schematic structural diagram of the audio SOC chip provided by the embodiment of the present application;

[0021] Figure 2 is a schematic flowchart of the audio playback method provided by the embodiment of the present application;

[0022] Figure 3 is a schematic diagram of the conversion principle of the DSM modulator provided by the embodiment of the present application;

[0023] Figure 4 is a schematic structural diagram of the 1-bit analog quantizer provided by the embodiment of the present application. Detailed Description of the Embodiments

[0024] To make the objectives, technical solutions, and advantages of the present application clearer, the following will further describe the embodiments of the present application in detail with reference to the drawings.

[0025] Figure 1 shows a schematic structural diagram of the audio SOC chip that can be applied to the present application.

[0026] As shown Figure 1 in the figure, the system architecture may include: a memory 1, an audio SOC chip 2, and a sound generating unit 3. The memory 1, the audio SOC chip 2, and the sound generating unit 3 may communicate with each other in a wired or wireless manner. For example, the wired communication link includes optical fiber, twisted pair, or coaxial cable, etc., and the wireless communication link includes a Bluetooth communication link, a Wireless-Fidelity (Wi-Fi) communication link, or a microwave communication link, etc.

[0027] Among them, the audio SOC chip 2 includes a digital filter 21, a DSM modulator 22, a 1-bit analog quantizer 23, and a power amplifier 24. The memory 1 is used to store the PCM audio digital sequence. The digital filter 21 is used to perform digital filtering processing. The DSM modulator 22 is used to convert N-bit sampling points into multiple 1-bit sampling points. The 1-bit analog quantizer 23 is used to convert 1-bit sampling points into 1-bit analog signals. The power amplifier 24 is used to perform power amplification on the analog signal. The sound generating unit 3 includes headphones, speakers, etc.

[0028] It should be understood Figure 1 that the numbers of the memory 1, the audio SOC chip 2, and the sound generating unit 3 in [[ ]] are only illustrative. According to the implementation requirements, it can be any number.

[0029] Please refer to Figure 2 , which is a schematic flowchart of an audio playback method provided by an embodiment of the present application. As shown Figure 2 in the figure, the method of the embodiment of the present application may include the following steps:

[0030] S201. Read the PCM audio digital sequence from the memory; the PCM audio digital sequence includes multiple N-bit sampling points, and N is an integer greater than 1.

[0031] Among them, the audio SOC chip reads the PCM audio digital sequence from the memory (such as Flash, DRAM, etc.) through its built-in or external memory interface. These sequences are composed of multiple N-bit sampling points, and each sampling point represents the amplitude value of the audio signal at a certain moment. N is an integer greater than 1, and is usually determined according to the trade-off between audio quality and storage space, such as common 16 bits, 24 bits, etc.

[0032] For example, if N is 16, then each sampling point is composed of 16 binary bits and can represent 65,536 different amplitude values. The audio SOC chip will read these sampling points in the storage order in sequence to prepare for subsequent conversion and processing.

[0033] Furthermore, in some embodiments of the present application, a digital filter is integrated inside the audio SOC chip. This digital filter can select an appropriate filter type according to application requirements, such as an IIR (Infinite Impulse Response) filter or an FIR (Finite Impulse Response) filter.

[0034] IIR filter: Implemented based on a recursive algorithm, it has high processing efficiency and low latency, but the phase response may not be linear and distortion is likely to occur. It is suitable for scenarios that require fast real-time signal processing, such as the preprocessing of audio signals.

[0035] FIR filter: Implemented based on a non-recursive algorithm, it has good stability, is easy to design, and has a controllable frequency response, but its structure is relatively complex and the latency is high. It is suitable for signal processing that requires high precision, such as noise removal and enhancing specific frequency components.

[0036] The audio SOC chip configures the digital filter according to preset filtering parameters (such as cut-off frequency, passband gain, stopband attenuation, etc.) and the selected filter type. These parameters are usually adjusted through register settings to meet different audio processing requirements.

[0037] Read the PCM audio digital sequence: The audio SOC chip reads the PCM audio digital sequence from the memory. This sequence contains multiple N-bit sampling points. The read PCM audio digital sequence is input into the digital filter module. The filter performs filtering processing on the input sequence according to the preset parameters and type, and outputs the processed PCM audio digital sequence. The processed PCM audio digital sequence is sent to the subsequent DSM modulator module for further processing.

[0038] Furthermore, in some embodiments of the present application, the audio SOC chip internally contains multiple registers for storing various configuration parameters, including filtering parameters. The chip accesses these registers through an internal bus or a specific interface to read the preset filtering parameters. The preset filtering parameters may include cut-off frequency, passband gain, stopband attenuation, etc. These parameters are written into the registers during the chip design stage or user configuration. After the chip reads the filtering parameters in the registers, it needs to parse these parameters. The parsing process may involve converting the binary values in the registers into actual filtering parameter values and understanding the meaning of these parameters according to the definitions in the chip specification or user manual.

[0039] An audio SOC chip usually contains multiple pre-set digital filters, and these filters have different characteristics, such as different cut-off frequencies, passband gains, stopband attenuations, etc. The chip selects the digital filter that is closest to the match in the internal filter library according to the read filtering parameters. To determine the most matching digital filter, the chip may adopt a matching algorithm. This algorithm compares the read filtering parameters with the parameters of each filter in the internal filter library and calculates the difference or similarity between them. Based on the calculation results of the difference or similarity, the algorithm selects the digital filter that is closest to the match. Once the matching digital filter is determined, the chip will configure this filter. The configuration process may involve setting the coefficients of the filter, adjusting the state of the filter, or enabling / disabling specific filter functions. After the configuration is completed, the chip will use the configured filter to process the input audio signal.

[0040] S202. Convert each N-bit sampling point into multiple 1-bit sampling points through a DSM modulator.

[0041] Among them, in the audio SOC chip, refer to Figure 3 the schematic diagram of the conversion principle shown. The function of the DSM (Delta-Sigma Modulation) modulator is to convert each high-precision N-bit PCM audio sampling point into a series of low-precision 1-bit sampling points. The key to this process lies in oversampling and noise shaping.

[0042] Oversampling means sampling the signal at a sampling rate higher than the Nyquist frequency. The Nyquist frequency is twice the highest frequency of the signal. By oversampling, the number of samples in the signal can be increased, thus providing more degrees of freedom for noise shaping. In the DSM modulator, the oversampling ratio (OSR) is usually much higher than the Nyquist sampling rate. For example, the OSR may be 64 times, 128 times or higher.

[0043] Noise shaping is another key feature of the DSM modulator. It uses a feedback loop to push the quantization noise (the noise generated due to the conversion from high precision to low precision) to higher frequencies. In this way, within the signal bandwidth (i.e., within the audio frequency range we care about), the quantization noise is significantly reduced, thus improving the signal-to-noise ratio (SNR).

[0044] When an N-bit sampling point enters the DSM modulator, it is first oversampled to generate a series of sampling points with shorter time intervals. Then, these oversampled points are processed through a noise shaping algorithm to generate a series of 1-bit sampling points. The number of these 1-bit sampling points depends on the oversampling ratio. For example, if the oversampling ratio is 64 times, then each N-bit sampling point may be converted into 64 or more 1-bit sampling points.

[0045] For example, there is a PCM audio digital sequence, where each sampling point is 16 bits (N = 16). To simplify the explanation, we select a sampling point in the sequence for demonstration. Assume the binary representation of this sampling point is 1000110000000000 (hexadecimal representation is 0x8C00).

[0046] Input sampling point: 1000110000000000 (16 bits). Principle of operation of the DSM modulator: The DSM modulator converts a high-precision digital signal into a series of low-precision 1-bit signals through oversampling and noise shaping techniques. In this example, we assume the oversampling rate is 64 times (i.e., each original sampling point is divided into 64 1-bit sampling points). The noise shaping technique pushes the quantization noise to the high-frequency band, thereby obtaining a higher signal-to-noise ratio in the low-frequency band.

[0047] The DSM modulator divides the input sampling point into 64 intervals according to its amplitude value (because the oversampling rate is 64). Each interval corresponds to a 1-bit output value (0 or 1). By comparing the amplitude value of the input sampling point with the boundary values of these intervals, the DSM modulator generates 64 1-bit sampling points.

[0048] Assume that after the processing of the DSM modulator, we obtain the following 1-bit sampling point sequence (only part of the sequence is listed here for simplicity): 1, 1, 1, 1, 1, 1, 1, 0, 0, 0,..., 0, 1 (a total of 64 1-bit sampling points). This sequence reflects the amplitude information of the original 16-bit sampling point 0x8C00. Although each 1-bit sampling point contains very little information (0 or 1), through oversampling and noise shaping techniques, the average value of these 1-bit sampling points will approximate the amplitude value of the original sampling point. Through the DSM modulator, we convert a 16-bit sampling point into 64 1-bit sampling points. These 1-bit sampling points together reflect the amplitude information of the original sampling point, and due to the effect of oversampling and noise shaping, they can provide a higher signal-to-noise ratio in the low-frequency band.

[0049] S203. Use a 1-bit analog quantizer to convert each 1-bit sampling point into a 1-bit analog signal.

[0050] Among them, the 1-bit analog quantizer in the audio SOC chip is responsible for converting each 1-bit sampling point into a 1-bit analog signal. This process is the inverse process of digitization, that is, converting discrete binary values into continuous analog signals.

[0051] A 1-bit quantizer usually realizes the conversion by comparing the sampling point with a fixed reference voltage. If the sampling point is 1, a high level is output; if it is 0, a low level is output. In this way, each 1-bit sampling point is converted into a corresponding analog level, forming a continuous 1-bit analog signal.

[0052] Furthermore, referring to Figure 4 the structural schematic diagram of the 1-bit analog quantizer shown, which includes a reference voltage circuit and a comparator with an inverting function.

[0053] The reference voltage circuit provides a fixed voltage reference VCC for the quantizer. This reference voltage is usually set to the middle value between the two levels (high level and low level) that the quantizer can recognize.

[0054] In a 1-bit quantizer, since there are only two possible output levels (0 and 1, or correspondingly, -V and +V, where V is the amplitude of the reference voltage), the design of the reference voltage circuit is crucial. It ensures that the quantizer can accurately compare the input signal with this reference and output the corresponding level.

[0055] The comparator is the core part of the quantizer and is responsible for comparing the input signal with the reference voltage.

[0056] In a 1-bit quantizer, the comparator usually has an inverting function, which means it can receive a positive input signal and an inverted reference voltage (or vice versa), and output a high level or a low level according to their relative magnitudes.

[0057] When the amplitude of the input signal is greater than the reference voltage, the comparator outputs a high level (e.g., +V); when the amplitude of the input signal is less than the reference voltage, the comparator outputs a low level (e.g., -V). In this way, each 1-bit sampling point is converted into the corresponding analog level.

[0058] S204. Use a power amplifier to amplify the 1-bit analog signal.

[0059] Among them, the analog signal after being converted by the 1-bit quantizer usually has a relatively low voltage and current level, which is not enough to directly drive the sound generating unit (such as a speaker or a headphone). Therefore, the audio SOC chip will use the built-in power amplifier (PA) to amplify the 1-bit analog signal.

[0060] The power amplifier increases the voltage and current of the signal so that it can drive the sound generating unit to produce a loud enough sound. During the amplification process, the power amplifier will keep the waveform and phase information of the signal unchanged to ensure the accurate transmission of the audio quality.

[0061] S205. Play the amplified 1-bit analog signal through the sound generating unit.

[0062] Finally, the audio SOC chip outputs the amplified 1-bit analog signal to the sound generating unit (such as a speaker or headphones), and sound is generated through its vibration. The sound generating unit vibrates according to the waveform of the input analog signal, generating air pressure waves. These waves propagate into the human ear and are received and analyzed by the auditory system, ultimately forming an auditory perception.

[0063] Since the PCM audio digital sequence has undergone precise sampling, quantization, and encoding processes, and through the effective conversion of the DSM modulator and 1-bit quantizer and the amplification process of the power amplifier, the finally played sound can maintain high sound quality and fidelity.

[0064] The embodiments of this application include the following beneficial effects:

[0065] Because high-performance DACs require complex circuit structures, including precise current sources, weight amplifiers, switch arrays, and output buffers, etc., these components occupy a relatively large chip area. In contrast, the DSM modulator and 1-bit analog quantizer in this solution have relatively simple structures, can effectively reduce the chip area occupation, thereby reducing costs and improving production efficiency. The DSM modulator converts each N-bit sampling point into multiple 1-bit sampling points, utilizing oversampling and noise shaping techniques, and can achieve high audio quality at a relatively low sampling rate. At the same time, the 1-bit analog quantizer directly converts each 1-bit sampling point into an analog signal, reducing information loss during the conversion process and improving the efficiency of audio conversion. This solution can also achieve high-quality audio playback by optimizing the circuit structure and signal processing flow. Especially the noise shaping ability of the DSM modulator can reduce in-band noise and improve the signal-to-noise ratio while ensuring audio quality, thereby enhancing the user's auditory experience. After adopting the DSM modulator and 1-bit analog quantizer, the overall design of the audio system becomes more concise and clear. This not only reduces the complexity of system design but also improves the reliability and stability of the system.

[0066] Those of ordinary skill in the art can understand that all or part of the processes in implementing the above-mentioned embodiment methods can be completed by instructing relevant hardware through a computer program. The program can be stored in a computer-readable storage medium. When the program is executed, it can include the processes of the above-mentioned method embodiments. Among them, the storage medium can be a magnetic disk, an optical disk, a read-only memory, or a random access memory, etc.

[0067] The above-disclosed are only the preferred embodiments of this application. Of course, it cannot be used to limit the scope of rights of this application. Therefore, equivalent changes made according to the claims of this application still fall within the scope covered by this application.

Claims

1. An audio playback method, characterized in that: include: Read PCM audio digital sequence from memory; The PCM audio digital sequence includes a plurality of N-bit sampling points, where N is an integer greater than 1; Each N-bit sampling point is converted into multiple 1-bit sampling points through a DSM modulator; Convert each 1-bit sampling point into a 1-bit analog signal using a 1-bit analog quantizer; A power amplifier is used to amplify the 1-bit analog signal. The amplified 1-bit analog signal is played through the sound unit.

2. The method according to claim 1, characterized in that Also includes: The read PCM audio digital sequence is filtered by a digital filter, and the filtered PCM audio digital sequence is sent to the DSM modulator.

3. The method according to claim 2, characterized in that Also includes: Reading preset filter parameters in the register, wherein the filter parameters include cutoff frequency, passband gain, and stopband attenuation; A matching digital filter is determined from a plurality of built-in digital filters according to the filtering parameters.

4. The method according to claim 3, characterized in that: The built-in multiple digital filters include: IIR filter and FIR filter.

5. The method according to claim 1 or 2 or 3 or 4, characterized in that: The 1-bit analog quantizer includes: a reference signal circuit and a comparator with an inversion function.

6. The method according to claim 5, characterized in that The memory is a flash memory.

7. The method according to claim 6, characterized in that The sampling rate of the DSM modulator is equal to 64 times or 128 times the Nyquist sampling rate.

8. An audio SOC chip, characterized in that: include: DSM modulator, used to read PCM audio digital sequence from memory; The PCM audio digital sequence includes a plurality of N-bit sampling points, where N is an integer greater than 1; each N-bit sampling point is converted into a plurality of 1-bit sampling points; A 1-bit analog quantizer, used to convert each 1-bit sampling point into a 1-bit analog signal; The power amplifier is used to amplify the 1-bit analog signal and play the amplified 1-bit analog signal through the sound unit.

9. The audio SOC chip according to claim 8, characterized in that: Also includes: The digital filter is used to filter the read PCM audio digital sequence and send the filtered PCM audio digital sequence to the DSM modulator.

10. An electronic device, characterized in that: include: A memory, an audio SOC chip as claimed in claim 8 or 9, and a sound unit.

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