Signal processing method and device, readable storage medium and computer program product

Abstract

The invention discloses a signal processing method and device, a readable storage medium and a computer program product, and relates to the technical field of signal processing, and the method comprises the steps: obtaining a to-be-processed original signal, and inputting the original signal to a high-pass filter; the duration of a low-frequency signal in the original signal is obtained, and the signal frequency of the low-frequency signal belongs to a preset low-frequency range; and adjusting filter parameters of the high-pass filter according to the duration, and determining a signal output by the high-pass filter as a processing result. According to the invention, the phenomenon of intermodulation distortion of the signal is reduced, and the audio quality is improved.

Description

Technical Field

[0001] The present application relates to the field of signal processing technology, and in particular to a signal processing method, device, readable storage medium and computer program product. Background Art

[0002] As users pursue thinner and lighter devices, the space occupied by the acoustic system in the device (including area and thickness) has been significantly reduced. The thickness of the speaker unit in the acoustic system is getting thinner and the amplitude is getting larger to obtain better low-frequency effects.

[0003] However, the signal output by the ultra-thin large-amplitude loudspeaker will have very serious nonlinear distortion, causing an increase in intermodulation distortion (IMD), resulting in turbid and unclear audio playback, the introduction of noise, and other problems that affect the audio quality.

[0004] Therefore, how to reduce the generation of signal intermodulation distortion to improve audio quality is a technical problem that needs to be solved urgently. Summary of the invention

[0005] The main purpose of this application is to provide a signal processing method, device, readable storage medium and computer program product, aiming at solving the technical problem of how to reduce the generation of signal intermodulation distortion to improve audio quality.

[0006] To achieve the above object, the present application provides a signal processing method, the signal processing method comprising:

[0007] Acquire an original signal to be processed, and input the original signal into a high-pass filter;

[0008] Obtaining the duration of a low-frequency signal in the original signal, wherein the low-frequency signal is a signal whose frequency belongs to a preset low-frequency range;

[0009] The filter parameters of the high-pass filter are adjusted according to the duration, and the signal output by the high-pass filter is determined as the processing result.

[0010] In one embodiment, the filter parameters include a cutoff frequency, and the step of adjusting the filter parameters of the high-pass filter according to the duration includes:

[0011] If the duration is greater than a preset duration, the cutoff frequency of the high-pass filter is adjusted to a first frequency, and the first frequency is greater than a lower limit value of the preset low-frequency range.

[0012] In one embodiment, the step of adjusting the filter parameters of the high-pass filter according to the duration includes:

[0013] If the duration is less than or equal to a preset duration, the cutoff frequency of the high-pass filter is adjusted to a second frequency, where the second frequency is less than the first frequency and the second frequency is less than an upper limit value of the preset low-frequency range.

[0014] In one embodiment, the filter parameters further include a filter order, and the step of adjusting the filter parameters of the high-pass filter according to the duration includes:

[0015] If the duration is greater than a preset duration, adjusting the filter order of the high-pass filter to a first order;

[0016] If the duration is less than or equal to a preset duration, the filter order of the high-pass filter is adjusted to a second order, wherein the first order is greater than the second order.

[0017] In one embodiment, after the step of obtaining the duration of the low-frequency signal in the original signal, the method further includes:

[0018] If the duration is less than or equal to the preset duration, the high-pass filter is controlled to enter a dormant state, and the step of determining that the signal output by the high-pass filter is a processing result is performed.

[0019] In one embodiment, the step of obtaining the duration of the low-frequency signal in the original signal includes:

[0020] Performing low-pass filtering on the original signal to obtain a low-frequency signal;

[0021] If the signal energy of the starting frame signal of the low-frequency signal is greater than or equal to a preset threshold, determining the sampling moment corresponding to the starting frame signal as the first moment, and traversing the next frame signal of the low-frequency signal in sequence;

[0022] Until the signal energy of the next frame signal traversed is less than the preset threshold, the sampling moment corresponding to the next frame signal traversed is determined as the second moment;

[0023] The duration between the first moment and the second moment is determined as the duration of the low-frequency signal.

[0024] In one embodiment, after the step of determining that the sampling moment corresponding to the low-frequency signal of the next frame to be traversed is the second moment, the method further includes:

[0025] If the next frame signal traversed is not the end frame signal of the low-frequency signal, continue to traverse the next frame signal of the low-frequency signal;

[0026] The next frame signal traversed is used as a new starting frame signal, and the process returns to the step of determining that the sampling moment corresponding to the starting frame signal is the first moment if the signal energy of the starting frame signal of the low-frequency signal is greater than or equal to a preset threshold, and traversing the next frame signal of the low-frequency signal in sequence until the ending frame signal of the low-frequency signal is traversed.

[0027] In addition, to achieve the above-mentioned purpose, the present application also provides a signal processing device, which includes: a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the computer program is configured to implement the steps of the signal processing method as described above.

[0028] In addition, to achieve the above-mentioned purpose, the present application also provides a readable storage medium, which is a computer-readable storage medium, and a program for implementing the signal processing method is stored on the computer-readable storage medium. The program for implementing the signal processing method is executed by a processor to implement the steps of the signal processing method as described above.

[0029] The present application also provides a computer program product, comprising a computer program, wherein when the computer program is executed by a processor, the steps of the signal processing method as described above are implemented.

[0030] One or more technical solutions proposed in this application have at least the following technical effects:

[0031] Obtain the original signal to be processed, and input the original signal into a high-pass filter; obtain the duration of the low-frequency signal in the original signal, wherein the low-frequency signal is a signal whose frequency belongs to a preset low-frequency range; adjust the filter parameters of the high-pass filter according to the duration, and determine that the signal output by the high-pass filter is the processing result. Considering that the intermodulation distortion of the signal mainly comes from the modulation of the high-frequency signal by the low-frequency signal, and the generation of intermodulation distortion mainly comes from the continuous low-frequency signal, because the continuous low-frequency signal will cause the medium and high frequencies to have a continuous amplitude envelope, which is easily perceived by the human ear; while the bursty low-frequency signal will only produce an instantaneous amplitude envelope, and it is difficult for the human ear to feel this kind of audio "jitter" in the changing audio. Thus, in the embodiment of the present application, after obtaining the original signal, the duration of the low-frequency signal in the original signal is obtained, and the filter parameters of the high-pass filter are adjusted according to the duration, so that when the duration is greater than the preset duration, the high-pass filter can filter out the low-frequency signal in the original signal by adjusting the filter parameters, thereby reducing or even avoiding the modulation of the low-frequency signal on the high-frequency signal, thereby reducing the generation of signal intermodulation distortion and improving the audio quality. In addition, the embodiment of the present application realizes the reduction of intermodulation distortion from the software level, without the need to design additional circuits or add additional hardware devices to reduce intermodulation distortion, thereby reducing the implementation cost of reducing intermodulation distortion. BRIEF DESCRIPTION OF THE DRAWINGS

[0032] The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate embodiments consistent with the present application and, together with the description, serve to explain the principles of the present application.

[0033] In order to more clearly illustrate the embodiments of the present application or the technical solutions in the prior art, the drawings required for use in the embodiments or the description of the prior art will be briefly introduced below. Obviously, for ordinary technicians in this field, other drawings can be obtained based on these drawings without paying any creative labor.

[0034] Figure 1 This is a flow chart of the first embodiment of the signal processing method of the present application;

[0035] Figure 2 A brief flowchart of the signal processing method of this application;

[0036] Figure 3 This is a schematic diagram of a signal duration detection process involved in an embodiment of a signal processing method of the present application;

[0037] Figure 4 A schematic diagram of a cutoff frequency-normalized amplitude variation curve of a high-pass filter involved in an embodiment of a signal processing method of the present application;

[0038] Figure 5 A schematic diagram of a filter order-normalized amplitude variation curve of a high-pass filter involved in an embodiment of a signal processing method of the present application;

[0039] Figure 6 This is a schematic diagram of the structure of the signal processing device of the present application;

[0040] Figure 7 Schematic diagram of the device structure of the hardware operating environment involved in the signal processing device in the embodiment of the present application.

[0041] The purpose, features and advantages of this application will be further described in conjunction with the embodiments and with reference to the accompanying drawings. DETAILED DESCRIPTION

[0042] In order to make the above-mentioned purposes, features and advantages of the present invention more obvious and easy to understand, the technical scheme in the embodiments of the present invention will be clearly and completely described below in conjunction with the drawings in the embodiments of the present invention. Obviously, the described embodiments are only part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without creative work belong to the scope of protection of the present invention.

[0043] With the development of extended reality technologies such as AR (Augmented Reality) and VR (Virtual Reality), the audio experience of head-mounted devices is increasingly valued by users. On the other hand, due to the pursuit of thinner and lighter devices, the space occupied by the acoustic system is increasingly tending to develop in the direction of large-area ultra-thin flattening, and the thickness of the speaker unit is getting thinner and the amplitude is getting larger to obtain better low-frequency effects. However, ultra-thin and large-amplitude speakers will have very serious nonlinear distortion, causing an increase in total harmonic distortion (THD) and intermodulation distortion (IMD), resulting in turbid and unclear audio playback, and introducing negative problems such as noise.

[0044] For head-mounted audio devices such as AR and VR, IMD is usually more likely to cause playback degradation. This is because intermodulation distortion comes from the modulation of high-frequency signals by low-frequency signals, which causes the entire high-frequency signal to have a low-frequency periodic envelope, causing the sound to tremble, resulting in turbidity and noise. Since these devices are close to the ear canal, low frequencies are easier to replay, and usually their low-frequency gain is larger, which also introduces more serious intermodulation distortion. Intermodulation distortion has exceeded the amplitude limit of the speaker and has become the biggest shortcoming that limits the performance of low frequencies.

[0045] Based on this, the main solution of the present application is: obtain the original signal to be processed, and input the original signal into a high-pass filter; obtain the duration of the low-frequency signal in the original signal, wherein the low-frequency signal is a signal whose frequency belongs to a preset low-frequency range; adjust the filter parameters of the high-pass filter according to the duration, and determine that the signal output by the high-pass filter is the processing result.

[0046] After obtaining the original signal, the present application obtains the duration of the low-frequency signal in the original signal, and adjusts the filter parameters of the high-pass filter according to the duration, so that when the duration is greater than the preset duration, the high-pass filter can filter out the low-frequency signal in the original signal by adjusting the filter parameters, thereby reducing or even avoiding the modulation of the low-frequency signal on the high-frequency signal, thereby reducing the generation of signal intermodulation distortion and improving the audio quality. In addition, the present application reduces the intermodulation distortion from the software level, without the need to design additional circuits or add additional hardware devices to reduce intermodulation distortion, thereby reducing the cost of reducing the intermodulation distortion.

[0047] It should be noted that the execution subject of each embodiment of the signal processing method of the present application may be a computing service device with data processing, network communication and program running functions, such as a tablet computer, a personal computer, a mobile phone, etc., or a signal processing device capable of realizing the above functions, such as an AR helmet, a VR helmet, AR glasses, VR glasses, headphones, etc. The embodiments of the signal processing method of the present application do not impose specific restrictions on this.

[0048] Based on this, the present application proposes a signal processing method of the first embodiment, please refer to Figure 1 , the signal processing method comprises steps S10 to S30:

[0049] Step S10, obtaining an original signal to be processed, and inputting the original signal into a high-pass filter;

[0050] It should be noted that, in order to facilitate the subsequent acquisition of duration, after acquiring the original signal, if the original signal is a frequency domain signal, the original signal in the frequency domain is converted to the time domain. Specifically, the frequency domain signal can be converted to the time domain by inverse Fourier transform, and subsequent processing is performed based on the original signal in the time domain.

[0051] After obtaining the original signal, the original signal can be input into a high-pass filter so that dynamic filtering of the original signal can be achieved by changing the filter parameters. Dynamic filtering of the original signal can be achieved through simple parameter adjustment, which is low-cost and easy to deploy in devices with lower computing power.

[0052] Step S20, obtaining the duration of the low-frequency signal in the original signal, wherein the low-frequency signal is a signal whose frequency belongs to a preset low-frequency range;

[0053] The low-frequency signal is a signal whose frequency belongs to the preset low-frequency range. It should be noted that relevant personnel can set the preset low-frequency range based on experience or experimental measurement. Generally speaking, when the signal contains both a low-frequency signal whose frequency belongs to the preset low-frequency range and a non-low-frequency signal that does not belong to the preset low-frequency range, the probability of the signal generating intermodulation distortion is greater than a certain value (the certain value can be set by relevant personnel based on actual needs, such as when the audio quality requirements are high, a smaller value can be set, and when the audio quality requirements are low, a larger value can be set). For example, in a specific implementation, the preset low-frequency range is less than or equal to 200 Hz.

[0054] The duration of the low-frequency signal may specifically be the duration between the time when the low-frequency signal begins to be detected in the original signal and the time when the low-frequency signal is no longer detected. For example, if the low-frequency signal is detected at time A of the original signal, and the low-frequency signal is always detected in the original signal after time A, until the low-frequency signal is no longer detected in the original signal at time B, then the duration of the low-frequency signal is the duration between time A and time B.

[0055] Step S30, adjusting the filter parameters of the high-pass filter according to the duration, and determining the signal output by the high-pass filter as the processing result.

[0056] Initially, the filter parameters of the high-pass filter can be set to preset initial parameter values. Relevant personnel can set the initial values ​​of the filter parameters according to actual needs, that is, relevant personnel can set preset initial parameter values ​​according to actual needs, and this embodiment does not make any specific restrictions on this.

[0057] The filter parameters include but are not limited to the cutoff frequency, and may also include the filter order. The cutoff frequency refers to the frequency at which the filter starts to significantly attenuate (reduce) the signal strength from allowing the signal to pass. It is a key parameter to describe the performance of the filter and determines the frequency range that the filter can pass. The order of the filter refers to the number of independent energy storage elements (such as capacitors, inductors, delay units, etc.) in the differential equation, transfer function or frequency response characteristics used in the filter design, or equivalently, it refers to the highest power of the denominator polynomial that constitutes the filter transfer function. The order of the filter determines the complexity and filtering characteristics of the filter, including the sharpness of its frequency response (i.e., the width of the transition band) and the attenuation rate in the stop band.

[0058] It should be noted that in order to realize the adjustable filter parameters of the high-pass filter, the high-pass filter can be a digital filter or an analog filter with adjustable parameters. The digital filter is a system that uses digital signal processing technology to realize the filtering function. It processes the digital signal to reduce or eliminate unwanted frequency components. Specifically, it can be implemented by software on a digital signal processor (DSP) or a microprocessor. An analog filter is an electronic filter that is used to process analog signals, i.e., continuously changing voltage or current signals. Its main function is to allow signals within a specific frequency range to pass while suppressing or reducing signals of other frequencies. Analog filters are usually composed of passive components such as resistors, capacitors, inductors, or active components such as operational amplifiers.

[0059] Further, in order to enable the high-pass filter to filter out the low-frequency signal in the original signal when the duration is greater than the preset duration to reduce the generation of signal intermodulation distortion, the filter parameters include a cutoff frequency, and the step of adjusting the filter parameters of the high-pass filter according to the duration includes:

[0060] Step S301: If the duration is greater than a preset duration, the cutoff frequency of the high-pass filter is adjusted to a first frequency, and the first frequency is greater than a lower limit of the preset low-frequency range.

[0061] Preferably, the first frequency is greater than or equal to the upper limit of the preset low-frequency range. In this way, after the original signal passes through the high-pass filter, the low-frequency signal in the preset low-frequency range can be partially or even completely filtered out (theoretically completely filtered out, but in practice may not achieve the ideal effect), thereby achieving the maximum effective filtering of the low-frequency signal in the original signal.

[0062] It should be noted that the preset duration may specifically be a duration set by relevant personnel based on experience or experimental measurements. Generally speaking, when the duration of a low-frequency signal is greater than the preset duration, the probability of the signal generating intermodulation distortion is greater than a certain value. The certain value may be set by relevant personnel based on actual needs. For example, when the audio quality requirement is high, a smaller value may be set; and when the audio quality requirement is low, a larger value may be set. This embodiment does not impose any specific restrictions on this.

[0063] Considering that intermodulation distortion causes changes in the amplitude envelope of the high-frequency signal, its envelope period is equal to half of the low-frequency period. When the low-frequency signal frequency is low, the envelope frequency is also lower, which is easier to be detected by the human ear. When the low-frequency signal frequency is high, the envelope frequency is larger, and the faster "jitter" is difficult for the human ear to detect. When the envelope frequency is greater than 220Hz, the human ear cannot detect the "jitter" of the sound, and thus cannot perceive the intermodulation distortion. Based on this, after detecting that the duration of the low-frequency signal in the original signal is greater than the preset duration, the low-frequency signal in the original signal is filtered out to eliminate the low-frequency signal, thereby avoiding the modulation of the higher-frequency high-frequency signal by the lower-frequency low-frequency signal, thereby avoiding the human ear's perception of intermodulation distortion.

[0064] Among them, the amplitude envelope (AE) refers to the amplitude change curve of an audio signal or a modulated signal over a period of time. It describes the change of signal strength over time. The envelope frequency refers to the frequency of the signal envelope (i.e. the changing profile of the signal amplitude). In many cases, especially when the signal is an amplitude modulated (AM) signal, the envelope will change over time, and the rate of this change is the envelope frequency.

[0065] In addition, considering that when the energy of high-frequency signals is high, their interaction in nonlinear systems is stronger, and thus intermodulation distortion is more likely to occur, and that intermodulation distortion mainly comes from the modulation of high-frequency signals by low-frequency signals, if low-frequency signals and high-frequency signals exist at the same time, the probability of intermodulation distortion will also be greatly increased. Based on this, if it is detected that the duration of the low-frequency signal in the original signal is less than or equal to the preset duration, the signal energy of the high-frequency signal in the original signal can be further detected, or whether the low-frequency signal and the high-frequency signal exist in the original signal at the same time. If the signal energy of the high-frequency signal is greater than the preset threshold, or the low-frequency signal and the high-frequency signal exist in the original signal at the same time, the low-frequency signal in the original signal is filtered out (filtered out by a high-pass filter, the same below, no further explanation will be given later). If the low-frequency signal and the high-frequency signal do not exist at the same time in the original signal and the signal energy of the high-frequency signal is less than or equal to the preset threshold, the original signal is not processed in any way to further reduce the generation of signal intermodulation distortion, thereby improving the audio quality.

[0066] Among them, the high-frequency signal is a signal whose signal frequency belongs to a preset high-frequency range, and the upper limit value of the preset low-frequency range is less than or equal to the lower limit value of the preset high-frequency range. For example, in a specific embodiment, the preset low-frequency range is less than or equal to 200 Hz, and the preset high-frequency range is 200 Hz to 1200 Hz.

[0067] In this embodiment, after obtaining the original signal, the duration of the low-frequency signal in the original signal is obtained, and the filter parameters of the high-pass filter are adjusted according to the duration, so that when the duration is greater than the preset duration, the high-pass filter can filter out the low-frequency signal in the original signal by adjusting the filter parameters, thereby reducing or even avoiding the modulation of the low-frequency signal on the high-frequency signal, thereby reducing the generation of signal intermodulation distortion and improving the audio quality. In addition, the embodiment of the present application realizes the reduction of intermodulation distortion from the software level, without the need to design additional circuits or add additional hardware devices to reduce intermodulation distortion, thereby reducing the implementation cost of reducing intermodulation distortion.

[0068] In order to solve the problem of intermodulation distortion, the usual method is to add a high-pass filter with a cutoff frequency greater than 110Hz to filter out the low-frequency signal in the signal. This is because intermodulation distortion causes the amplitude envelope of the high frequency, and its envelope period is equal to half of the low-frequency period. When the low-frequency signal frequency is low, the envelope frequency is also lower, which is easier to be detected by the human ear. When the low-frequency signal frequency is high, the envelope frequency is larger, and the faster "jitter" is difficult for the human ear to detect. According to psychoacoustic research, when the envelope frequency is greater than 220Hz, the human ear cannot detect the "jitter" of the sound, and thus cannot perceive intermodulation distortion. Adding a high-pass filter with a cutoff frequency greater than 110Hz eliminates low-frequency signals below the cutoff frequency, thereby avoiding the modulation of the higher-frequency high-frequency signal by the lower-frequency low-frequency signal, and thus avoiding the human ear's perception of intermodulation distortion.

[0069] However, the above operation will also introduce negative effects. From a subjective listening point of view, the frequency of 40 Hz to 80 Hz will significantly affect the diving depth and atmosphere of low-frequency drum beats. There are also a lot of low-frequency components from 80 Hz to 110 Hz. Adding a high-pass filter with a cutoff frequency greater than 110 Hz will significantly affect the low-frequency performance of the signal. It can be said that in order to solve the problem of intermodulation distortion, you can only choose between low-frequency volume and distortion, and it is difficult to have both.

[0070] Based on this and the first embodiment of the present application, in the second embodiment of the present application, the same or similar contents as those in the above-mentioned embodiment 1 can be referred to the above introduction, and no further description will be given later. On this basis, the step of adjusting the filter parameters of the high-pass filter according to the duration includes:

[0071] Step A20, if the duration is less than or equal to the preset duration, the cutoff frequency of the high-pass filter is adjusted to a second frequency, wherein the second frequency is less than the first frequency and the second frequency is less than the upper limit of the preset low-frequency range.

[0072] The second frequency is less than the first frequency, and the second frequency is less than the upper limit of the preset low frequency range to ensure that the low frequency signal can be retained after passing through the high pass filter. Preferably, the second frequency can be less than or equal to the lower limit of the preset low frequency range and greater than or equal to zero to retain as much low frequency signal as possible (theoretically, it is completely retained at this time) to improve the low frequency volume.

[0073] In this embodiment, if it is detected that the duration of the low-frequency signal in the original signal is less than or equal to the preset duration, the cutoff frequency of the high-pass filter is lowered to retain part or all of the low-frequency signal. In this way, when it is detected that the duration of the low-frequency signal in the original signal is less than or equal to the preset duration, the low-frequency signal in the original signal is retained, which can improve the low-frequency volume of the signal. When it is detected that the duration of the low-frequency signal in the original signal is greater than the preset duration, the low-frequency signal in the original signal will be filtered out, which can reduce the occurrence of intermodulation distortion, thereby achieving both low-frequency volume and reduced intermodulation distortion.

[0074] In a possible implementation manner, after the step of obtaining the duration of the low-frequency signal in the original signal, the method further includes:

[0075] Step B10: If the duration is less than or equal to a preset duration, the high-pass filter is controlled to enter a dormant state, and the step of determining that the signal output by the high-pass filter is a processing result is executed.

[0076] It should be noted that the sleep state refers to the state in which the high-pass filter is not working, that is, the high-pass filter is not started. At this time, the high-pass filter can be understood as a data transmission channel, and no processing is performed on the input data. In this embodiment, no processing is performed on the original signal.

[0077] If it is detected that the duration of the low-frequency signal in the original signal is less than or equal to the preset duration, the high-pass filter is controlled to enter a dormant state. In this way, the next time the duration of the low-frequency signal in the original signal is detected to be greater than the preset duration, the high-pass filter can be directly started without updating the filter parameters of the high-pass filter again, thereby reducing the number of times the filter parameters of the high-pass filter are updated.

[0078] Based on the first embodiment and / or the second embodiment of the present application, in the third embodiment of the present application, the same or similar contents as those in the first and second embodiments can be referred to the above description, and will not be described in detail later. On this basis, the filter parameters also include the filter order, and the step of adjusting the filter parameters of the high-pass filter according to the duration includes:

[0079] Step C10, if the duration is greater than a preset duration, adjusting the filter order of the high-pass filter to a first order;

[0080] Step C20: If the duration is less than or equal to a preset duration, adjusting the filter order of the high-pass filter to a second order, wherein the first order is greater than the second order.

[0081] It is understandable that a high-order filter with a higher filter order has a sharper cutoff characteristic. This means that near the cutoff frequency, the filter can drop from the passband to the stopband faster, providing better frequency selectivity, while a low-order filter with a lower filter order has a relatively flat cutoff characteristic, and a wider transition area between the passband and the stopband. In this way, when it is detected that the duration of the low-frequency signal in the original signal is greater than the preset duration, the filter order of the high-pass filter is adjusted to a higher first order, which can better filter out the low-frequency signal in the original signal to further reduce the generation of intermodulation distortion.

[0082] Based on the first embodiment, the second embodiment and / or the third embodiment of the present application, in the fourth embodiment of the present application, the same or similar contents as those in the first embodiment, the second embodiment and the third embodiment can be referred to the above introduction, and will not be described in detail later. On this basis, the step of obtaining the duration of the low-frequency signal in the original signal includes:

[0083] Step E10, performing low-pass filtering on the original signal to obtain a low-frequency signal;

[0084] The original signal is low-pass filtered to obtain a low-frequency signal whose signal frequency is within a preset low-frequency range. Specifically, the original signal can be low-pass filtered by a low-pass filter, and relevant personnel can also set other low-pass filtering methods, which are not specifically limited in this embodiment.

[0085] After obtaining the low-frequency signal, the low-frequency signal may be subjected to framing processing to divide the low-frequency signal into multiple frame signals. It should be noted that if the original signal has been subjected to framing processing, the frame length of the obtained low-frequency signal may be consistent with the frame length of the original signal, and there is no need to perform repeated framing processing on the low-frequency signal.

[0086] Step E20, if the signal energy of the starting frame signal of the low-frequency signal is greater than or equal to a preset threshold, determining the sampling moment corresponding to the starting frame signal as the first moment, and traversing the next frame signal of the low-frequency signal in sequence;

[0087] It can be understood that the audio signal heard by the human ear is an analog signal. When collecting analog signals, the signals need to be sampled, quantized and encoded before they can finally be stored or transmitted. Specifically, sampling is the process of converting continuous analog signals into discrete signals. By sampling the signal at fixed time intervals, a series of sample values ​​can be obtained. Therefore, the collected audio signal consists of one or more signal sampling points.

[0088] The start frame signal may be the first frame signal of the low frequency signal. For a frame signal, the sampling time corresponding to the signal may be the same sampling time as the first signal sampling point of the frame signal, that is, the start time of the frame signal.

[0089] Further, the time corresponding to the first signal sampling point of the start frame signal can be set as time zero, so as to obtain the time corresponding to each signal sampling point in the signal (i.e., sampling time) with time zero as the starting time. It is easy to understand that the time corresponding to each signal sampling point in the signal can be obtained based on a known method, such as the time corresponding to the signal sampling point can be obtained based on the signal sampling rate and the index of the signal sampling point. This embodiment does not describe in detail the method for obtaining the sampling time corresponding to the signal sampling point.

[0090] It should be noted that, for a certain frame signal, the signal energy of the frame signal may specifically be the average energy of the frame signal. The preset threshold may specifically be a value set in advance by relevant personnel, and this embodiment does not impose any specific limitation on this.

[0091] It should be noted that if the signal energy of the starting frame signal is less than the preset threshold, the next frame signal of the low-frequency signal will continue to be traversed in sequence until the signal energy of the next frame signal is greater than or equal to the preset threshold, then the sampling moment corresponding to the next frame signal is determined to be the first moment, and the next frame signal of the low-frequency signal will continue to be traversed until the signal energy of the next frame signal is less than the preset threshold, then the sampling moment corresponding to the next frame signal is determined to be the second moment.

[0092] Step E30, until the signal energy of the next frame signal traversed is less than a preset threshold, then determining the sampling time corresponding to the next frame signal traversed as the second time;

[0093] Step E40: Determine the duration between the first moment and the second moment as the duration of the low-frequency signal.

[0094] The duration between the first moment and the second moment is determined as the duration, and after the duration is obtained, subsequent steps are performed according to the duration, such as adjusting a filter parameter of a high-pass filter or changing a working state of the high-pass filter according to the duration.

[0095] Furthermore, the filtering process of the high-pass filter and the detection of the low-frequency signal in the original signal can be performed synchronously. For example, initially, the high-pass filter filters the original signal with a preset initial parameter value, until a second moment detects that the duration of the low-frequency signal is greater than the preset duration, then the cutoff frequency of the high-pass filter is adjusted to the first frequency, and the high-pass filter filters the original signal after the second moment with the first frequency until the next change of the filter parameters. For the original signal between the initial moment and the second moment, the high-pass filter still filters the original signal within this period with the preset initial parameter value. Similarly, if the duration of the low-frequency signal is less than or equal to the preset duration at the second moment, the cutoff frequency of the high-pass filter is adjusted to the second frequency or the high-pass filter enters a dormant state, and the high-pass filter filters the original signal after the second moment with the second frequency or does not perform any processing on the original signal after the second moment until the next change of the filter parameters.

[0096] In a possible implementation manner, after the step of determining that the sampling moment corresponding to the low-frequency signal of the next frame to be traversed is the second moment, the method further includes:

[0097] Step F10, if the next frame signal traversed is not the end frame signal of the low-frequency signal, continue to traverse the next frame signal of the low-frequency signal;

[0098] The end frame signal may specifically be the last frame signal of the low-frequency signal.

[0099] Step F20, taking the next frame signal traversed as a new starting frame signal, returning to execute the step of determining the sampling moment corresponding to the starting frame signal as the first moment if the signal energy of the starting frame signal of the low-frequency signal is greater than or equal to the preset threshold, and traversing the next frame signal of the low-frequency signal in sequence until the end frame signal of the low-frequency signal is traversed.

[0100] It can be understood that for each first moment, a second moment after the first moment (considered to be the second moment corresponding to the first moment) and the duration of the two moments can be obtained, and the corresponding signal processing strategy (filtering out low-frequency signals or retaining low-frequency signals) can be obtained by obtaining the duration. Then, the signal processing strategy can be used to process the signal between the previous first moment (the first first moment, which is the moment corresponding to the initial signal sampling point) and the first moment, or the signal processing strategy can be used to process the signal between the first moment and the next first moment (the last first moment, which is the moment of the end signal sampling point pair), or the signal processing strategy can be used to process the signal between the previous second moment and the second moment corresponding to the first moment, or the signal processing strategy can be used to process the signal between the second moment corresponding to the first moment and the next second moment, or the relevant personnel can also set other processing methods to ensure that each segment of the original signal has a unique corresponding signal processing strategy.

[0101] For example, to help understand the technical concept or technical principle of the signal processing method after the present embodiment is combined with the first embodiment, the second embodiment and the third embodiment, please refer to Figure 2 As shown, the signal processing flow is:

[0102] Get the original signal ( Figures 2 to 3 After the input signal shown in the figure is obtained, the original signal is copied to obtain two copies of the original signal, one of which is input into the high-pass filter, and the other is input into the signal detector. The signal detector performs pattern recognition on the original signal according to the signal characteristics to obtain a recognition result, and determines an adjustment strategy for the filter parameters based on the recognition result according to the set logic, so as to adjust the filter parameters of the high-pass filter based on the adjustment strategy, and filter the signal through the high-pass filter to achieve the purpose of reducing or eliminating intermodulation distortion.

[0103] Specifically, refer to Figures 3 to 5 As shown, the signal detector is used to increase the cutoff frequency Fc and the stopband attenuation rate of the high-pass filter (by dynamically adjusting the filter order) to filter out the low-frequency signal in the original signal if it is detected that the duration of the low-frequency signal in the original signal is greater than the preset duration, that is, the low-frequency signal is a continuous low frequency. If it is detected that the duration of the low-frequency signal in the original signal is less than or equal to the preset duration, that is, the low-frequency signal is a bursty low frequency, the high-pass filter will not be started to retain the low-frequency signal in the original signal.

[0104] Furthermore, Figure 4 This is a curve diagram of the change of the normalized amplitude of the high-pass filter with the cutoff frequency. The horizontal axis represents the frequency and the vertical axis represents the normalized amplitude of the filter. Figure 5This is a curve of the change of the normalized amplitude of the high-pass filter with the stopband attenuation rate (that is, the filter order). The horizontal axis represents the frequency and the vertical axis represents the normalized amplitude of the filter. Among them, the normalized amplitude of the filter is a relative measure, which is used to describe the response of the filter to signals of different frequencies, rather than using absolute amplitude units (such as volts or decibels). At the cutoff frequency Fc and above Fc, the normalized amplitude of the filter is 1 (or 0dB), which means that the output amplitude is equal to the input amplitude. Below the cutoff frequency, the normalized amplitude is less than 1 (or a negative decibel value), indicating that the signal is attenuated.

[0105] It should be noted that the above examples are only used to assist in understanding the present application and do not constitute a limitation on the signal processing method of the present application. More simple transformations based on this technical concept are all within the scope of protection of the present application.

[0106] In addition, to achieve the above-mentioned purpose, the embodiment of the present application also provides a signal processing device, referring to Figure 6 As shown, the signal processing device includes:

[0107] An acquisition module 10 is used to acquire an original signal to be processed and input the original signal into a high-pass filter;

[0108] A detection module 20, used to obtain the duration of the low-frequency signal in the original signal, wherein the low-frequency signal is a signal whose frequency belongs to a preset low-frequency range;

[0109] The adjustment module 30 is used to adjust the filter parameters of the high-pass filter according to the duration, and determine the signal output by the high-pass filter as the processing result.

[0110] In one embodiment, the filter parameters include a cutoff frequency, and the adjustment module 30 is further configured to:

[0111] If the duration is greater than a preset duration, the cutoff frequency of the high-pass filter is adjusted to a first frequency, and the first frequency is greater than a lower limit value of the preset low-frequency range.

[0112] In one embodiment, the adjustment module 30 is further used for:

[0113] If the duration is less than or equal to a preset duration, the cutoff frequency of the high-pass filter is adjusted to a second frequency, where the second frequency is less than the first frequency and the second frequency is less than an upper limit value of the preset low-frequency range.

[0114] In one embodiment, the filter parameters further include a filter order, and the adjustment module 30 is further configured to:

[0115] If the duration is greater than a preset duration, adjusting the filter order of the high-pass filter to a first order;

[0116] If the duration is less than or equal to a preset duration, the filter order of the high-pass filter is adjusted to a second order, wherein the first order is greater than the second order.

[0117] In one embodiment, the signal processing device further includes a sleep module, and the sleep module is used to:

[0118] If the duration is less than or equal to the preset duration, the high-pass filter is controlled to enter a dormant state, and the step of determining that the signal output by the high-pass filter is a processing result is performed.

[0119] In one embodiment, the detection module 20 is further used for:

[0120] Performing low-pass filtering on the original signal to obtain a low-frequency signal;

[0121] If the signal energy of the starting frame signal of the low-frequency signal is greater than or equal to a preset threshold, determining the sampling moment corresponding to the starting frame signal as the first moment, and traversing the next frame signal of the low-frequency signal in sequence;

[0122] Until the signal energy of the next frame signal traversed is less than the preset threshold, the sampling moment corresponding to the next frame signal traversed is determined as the second moment;

[0123] The duration between the first moment and the second moment is determined as the duration of the low-frequency signal.

[0124] In one embodiment, the detection module 20 is further used for:

[0125] If the next frame signal traversed is not the end frame signal of the low-frequency signal, continue to traverse the next frame signal of the low-frequency signal;

[0126] Taking the traversed next frame signal as a new start frame signal, if the traversed next frame signal is not the end frame signal of the low-frequency signal, then continuing to traverse the next frame signal of the low-frequency signal;

[0127] The next frame signal traversed is used as a new starting frame signal.

[0128] In addition, an embodiment of the present application also proposes a signal processing device, which includes: a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the computer program is configured to implement the steps of the signal processing method as described above.

[0129] In addition, reference Figure 7 , which shows a schematic diagram of the structure of a signal processing device suitable for implementing an embodiment of the present application. The signal processing device in the embodiment of the present application may also include but is not limited to mobile terminals such as AR helmets, VR helmets, headphones, AR glasses, VR glasses, mobile phones, laptops, digital broadcast receivers, PDAs (Personal Digital Assistants), PADs (Portable Application Descriptions), PMPs (Portable Media Players), etc., and fixed terminals such as digital TVs, desktop computers, etc. Figure 7 The signal processing device shown is merely an example and should not bring any limitation to the functions and scope of use of the embodiments of the present application.

[0130] like Figure 7 As shown, the signal processing device may include a processing device 1001 (e.g., a central processing unit, a graphics processor, etc.), which can perform various appropriate actions and processes according to a program stored in a read-only memory (ROM: Read Only Memory) 1002 or a program loaded from a storage device 1003 to a random access memory (RAM: Random Access Memory) 1004. In RAM1004, various programs and data required for the operation of the signal processing device are also stored. The processing device 1001, ROM1002, and RAM1004 are connected to each other through a bus 1005. An input / output (I / O) interface 1006 is also connected to the bus. Generally, the following systems can be connected to the I / O interface 1006: an input device 1007 including, for example, a touch screen, a touch pad, a keyboard, a mouse, an image sensor, a microphone, an accelerometer, a gyroscope, etc.; an output device 1008 including, for example, a liquid crystal display (LCD: Liquid Crystal Display), a speaker, a vibrator, etc.; a storage device 1003 including, for example, a magnetic tape, a hard disk, etc.; and a communication device 1009. The communication device 1009 can allow the signal processing device to communicate with other devices wirelessly or by wire to exchange data. Although the figure shows a signal processing device with various systems, it should be understood that it is not required to implement or have all the systems shown. More or fewer systems can be implemented or provided instead.

[0131] In particular, according to the embodiments disclosed in the present application, the process described above with reference to the flowchart can be implemented as a computer software program. For example, the embodiments disclosed in the present application include a computer program product, which includes a computer program carried on a computer-readable medium, and the computer program includes a program code for executing the method shown in the flowchart. In such an embodiment, the computer program can be downloaded and installed from a network through a communication device, or installed from a storage device 1003, or installed from a ROM 1002. When the computer program is executed by the processing device 1001, the above-mentioned functions defined in the method of the embodiment disclosed in the present application are executed.

[0132] The signal processing device provided in the embodiment of the present application adopts the signal processing method in the above embodiment to solve the problem of how to reduce the generation of signal intermodulation distortion to improve the audio quality. Compared with the prior art, the beneficial effects of the signal processing device provided in the present application are the same as the beneficial effects of the signal processing method provided in the above embodiment, and other technical features in the signal processing device are the same as the features disclosed in the method of the previous embodiment, which will not be repeated here.

[0133] It should be understood that the various parts disclosed in this application can be implemented by hardware, software, firmware or a combination thereof. In the description of the above embodiments, specific features, structures, materials or characteristics can be combined in any one or more embodiments or examples in a suitable manner.

[0134] The above is only a specific implementation of the present application, but the protection scope of the present application is not limited thereto. Any person skilled in the art who is familiar with the present technical field can easily think of changes or substitutions within the technical scope disclosed in the present application, which should be included in the protection scope of the present application. Therefore, the protection scope of the present application should be based on the protection scope of the claims.

[0135] In addition, to achieve the above-mentioned purpose, an embodiment of the present application also provides a readable storage medium having computer-readable program instructions (ie, computer program) stored thereon, and the computer-readable program instructions are used to execute the signal processing method in the above-mentioned embodiment.

[0136] The computer-readable storage medium provided in the embodiment of the present application may be, for example, a USB flash drive, but is not limited to electrical, magnetic, optical, electromagnetic, infrared, or semiconductor systems, systems or devices, or any combination of the above. More specific examples of computer-readable storage media may include, but are not limited to: an electrical connection with one or more wires, a portable computer disk, a hard disk, a random access memory (RAM: Random Access Memory), a read-only memory (ROM: Read Only Memory), an erasable programmable read-only memory (EPROM: Erasable Programmable Read Only Memory or flash memory), an optical fiber, a portable compact disk read-only memory (CD-ROM: CD-Read Only Memory), an optical storage device, a magnetic storage device, or any suitable combination of the above. In this embodiment, the computer-readable storage medium may be any tangible medium containing or storing a program that can be used by or in combination with an instruction execution system, system or device. The program code contained on the computer-readable storage medium may be transmitted using any appropriate medium, including but not limited to: wires, optical cables, RF (Radio Frequency: Radio Frequency), etc., or any suitable combination of the above.

[0137] The computer-readable storage medium may be included in the signal processing device; or may exist independently without being assembled into the signal processing device.

[0138] The computer-readable storage medium carries one or more programs. When the one or more programs are executed by a signal processing device, the signal processing device: obtains an original signal to be processed, and inputs the original signal into a high-pass filter; obtains the duration of a low-frequency signal in the original signal, wherein the low-frequency signal is a signal whose frequency belongs to a preset low-frequency range; adjusts the filter parameters of the high-pass filter according to the duration, and determines that the signal output by the high-pass filter is the processing result.

[0139] Computer program code for performing the operations of the present application may be written in one or more programming languages ​​or a combination thereof, including object-oriented programming languages ​​such as Java, Smalltalk, C++, and conventional procedural programming languages ​​such as "C" or similar programming languages. The program code may be executed entirely on the user's computer, partially on the user's computer, as a separate software package, partially on the user's computer and partially on a remote computer, or entirely on a remote computer or server. In the case of a remote computer, the remote computer may be connected to the user's computer through any type of network, including a local area network (LAN) or a wide area network (WAN), or may be connected to an external computer (e.g., via the Internet using an Internet service provider).

[0140] The flow chart and block diagram in the accompanying drawings illustrate the possible architecture, function and operation of the system, method and computer program product according to various embodiments of the present application. In this regard, each square box in the flow chart or block diagram can represent a module, a program segment or a part of a code, and the module, the program segment or a part of the code contains one or more executable instructions for realizing the specified logical function. It should also be noted that in some alternative implementations, the functions marked in the square box can also occur in a sequence different from that marked in the accompanying drawings. For example, two square boxes represented in succession can actually be executed substantially in parallel, and they can sometimes be executed in the opposite order, depending on the functions involved. It should also be noted that each square box in the block diagram and / or flow chart, and the combination of the square boxes in the block diagram and / or flow chart can be implemented with a dedicated hardware-based system that performs a specified function or operation, or can be implemented with a combination of dedicated hardware and computer instructions.

[0141] The modules involved in the embodiments described in this application may be implemented by software or hardware, wherein the name of the module does not constitute a limitation on the unit itself in some cases.

[0142] The readable storage medium provided by the present application is a computer-readable storage medium, which stores computer-readable program instructions (i.e., computer programs) for executing the above-mentioned signal processing method, and can solve how to reduce the generation of signal intermodulation distortion to improve audio quality. Compared with the prior art, the beneficial effects of the computer-readable storage medium provided by the present application are the same as the beneficial effects of the signal processing method provided by the above-mentioned embodiment, and will not be repeated here.

[0143] In addition, an embodiment of the present application further proposes a computer program product, including a signal processing program, which implements the steps of the signal processing method described above when executed by a processor.

[0144] The specific implementation of the computer program product of the present application is basically the same as the above-mentioned signal processing method embodiments, and will not be repeated here.

[0145] It should be noted that, in this article, the terms "include", "comprises" or any other variations thereof are intended to cover non-exclusive inclusion, so that a process, method, article or system including a series of elements includes not only those elements, but also other elements not explicitly listed, or also includes elements inherent to such process, method, article or system. In the absence of further restrictions, an element defined by the sentence "comprises a ..." does not exclude the existence of other identical elements in the process, method, article or system including the element.

[0146] The serial numbers of the above-mentioned embodiments of the present application are for description only and do not represent the advantages or disadvantages of the embodiments.

[0147] Through the description of the above implementation methods, those skilled in the art can clearly understand that the above-mentioned embodiment methods can be implemented by means of software plus a necessary general hardware platform, and of course by hardware, but in many cases the former is a better implementation method. Based on this understanding, the technical solution of the present application is essentially or the part that contributes to the prior art can be embodied in the form of a software sensor, which is stored in a storage medium (such as ROM / RAM, disk, CD) as described above, including a number of instructions for a terminal device (which can be a mobile phone, computer, server, air conditioner, or network device, etc.) to execute the methods described in each embodiment of the present application.

[0148] The above are only preferred embodiments of the present application, and are not intended to limit the patent scope of the present application. Any equivalent structure or equivalent process transformation made using the contents of the present application specification and drawings, or directly or indirectly applied in other related technical fields, are also included in the patent protection scope of the present application.

Claims

1. A signal processing method, characterized in that: The signal processing method comprises the following steps: Acquire an original signal to be processed, and input the original signal into a high-pass filter; Obtaining the duration of a low-frequency signal in the original signal, wherein the low-frequency signal is a signal whose frequency belongs to a preset low-frequency range; The filter parameters of the high-pass filter are adjusted according to the duration, and the signal output by the high-pass filter is determined as the processing result.

2. The signal processing method according to claim 1, characterized in that: The filter parameters include a cutoff frequency, and the step of adjusting the filter parameters of the high-pass filter according to the duration includes: If the duration is greater than a preset duration, the cutoff frequency of the high-pass filter is adjusted to a first frequency, and the first frequency is greater than a lower limit value of the preset low-frequency range.

3. The signal processing method according to claim 2, characterized in that: The step of adjusting the filter parameters of the high-pass filter according to the duration includes: If the duration is less than or equal to a preset duration, the cutoff frequency of the high-pass filter is adjusted to a second frequency, where the second frequency is less than the first frequency and the second frequency is less than an upper limit of the preset low-frequency range.

4. The signal processing method according to claim 2, characterized in that: The filter parameters also include a filter order, and the step of adjusting the filter parameters of the high-pass filter according to the duration includes: If the duration is greater than a preset duration, adjusting the filter order of the high-pass filter to a first order; If the duration is less than or equal to a preset duration, the filter order of the high-pass filter is adjusted to a second order, wherein the first order is greater than the second order.

5. The signal processing method according to claim 1, characterized in that: After the step of obtaining the duration of the low-frequency signal in the original signal, the method further includes: If the duration is less than or equal to the preset duration, the high-pass filter is controlled to enter a dormant state, and the step of determining that the signal output by the high-pass filter is a processing result is performed.

6. The signal processing method according to claim 1, characterized in that: The step of obtaining the duration of the low-frequency signal in the original signal comprises: Performing low-pass filtering on the original signal to obtain a low-frequency signal; If the signal energy of the starting frame signal of the low-frequency signal is greater than or equal to a preset threshold, determining the sampling moment corresponding to the starting frame signal as the first moment, and traversing the next frame signal of the low-frequency signal in sequence; Until the signal energy of the next frame signal traversed is less than the preset threshold, the sampling moment corresponding to the next frame signal traversed is determined as the second moment; The duration between the first moment and the second moment is determined as the duration of the low-frequency signal.

7. The signal processing method according to claim 6, characterized in that: After the step of determining that the sampling moment corresponding to the low-frequency signal of the next frame to be traversed is the second moment, the method further includes: If the next frame signal traversed is not the end frame signal of the low-frequency signal, continue to traverse the next frame signal of the low-frequency signal; The next frame signal traversed is used as a new starting frame signal, and the process returns to the step of determining that the sampling moment corresponding to the starting frame signal is the first moment if the signal energy of the starting frame signal of the low-frequency signal is greater than or equal to a preset threshold, and traversing the next frame signal of the low-frequency signal in sequence until the ending frame signal of the low-frequency signal is traversed.

8. A signal processing device, characterized in that: The signal processing device comprises: a memory, a processor, and a computer program stored in the memory and executable on the processor, wherein the computer program is configured to implement the steps of the signal processing method according to any one of claims 1 to 7.

9. A readable storage medium, characterized in that: The readable storage medium is a computer-readable storage medium, and a computer program is stored on the readable storage medium. When the computer program is executed by a processor, the steps of the signal processing method according to any one of claims 1 to 7 are implemented.

10. A computer program product, characterized in that The computer program product comprises a computer program, and when the computer program is executed by a processor, the steps of the signal processing method according to any one of claims 1 to 7 are implemented.

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