Digital audio true peak limiting method, apparatus, device, and storage medium

Abstract

The application discloses a digital audio true peak limiting method and device, equipment and storage medium, and relates to the technical field of audio processing. The method comprises the following steps: acquiring an audio sampling signal, performing interpolation operation on the audio sampling signal in a preset reading window to obtain an interpolation result, and screening a true peak detection result; converting a previous time envelope signal into a current signal level, determining a release time coefficient according to the current signal level and a preset limiting threshold, and generating a current time envelope signal according to the true peak detection result and the release time coefficient; determining a target gain value according to the current time envelope signal and the preset limiting threshold; determining a target sampling point signal from the audio sampling signal, and obtaining a limiting audio signal according to the target gain value and the target sampling point signal. The application can improve the output sound quality of the audio signal after limiting processing.

Description

Technical Field

[0001] This application relates to the field of audio processing technology, and in particular to a digital audio true peak limiting method, apparatus, device, and storage medium. Background Technology

[0002] In the field of digital audio processing, limiters are a key module for preventing signal overload. Current limiters typically rely on peak detection at sampling points, applying attenuation gain if the signal exceeds a preset threshold. However, existing solutions only detect the amplitude of discrete sampling points, failing to reflect the true peak value in the analog domain after digital-to-analog conversion, and are prone to implicit clipping. Furthermore, their gain release time is a fixed value, which cannot simultaneously address transient response and background noise stability when the signal has a large dynamic range. Therefore, improving the output sound quality of the audio signal after limiter processing remains a problem that needs to be solved.

[0003] The above content is only used to help understand the technical solution of this application and does not represent an admission that the above content is prior art. Summary of the Invention

[0004] The main objective of this application is to provide a digital audio true peak limiting method, apparatus, device, and storage medium, aiming to solve the technical problem of how to improve the output sound quality of audio signals after limiting processing.

[0005] To achieve the above objectives, this application proposes a digital audio true peak limiting method, the method comprising: Acquire an audio sampling signal, perform interpolation operations on discrete sampling points in the audio sampling signal within a preset reading window to obtain interpolation results, and filter out true peak detection results from the interpolation results; The envelope signal from the previous moment is converted into the current signal level. The release time coefficient is determined based on the current signal level and a preset amplitude limiting threshold. Envelope tracking is performed based on the true peak detection result and the release time coefficient to generate the envelope signal at the current moment. An initial gain value is determined based on the current envelope signal and the preset amplitude limiting threshold, and the variation range of the initial gain value is limited to obtain the target gain value. The target sampling point signal is determined from the audio sampling signal, and a limited audio signal is obtained based on the target gain value and the target sampling point signal. The sampling point of the target sampling point signal is determined by the preset reading window.

[0006] In one embodiment, the step of performing interpolation operations on discrete sampling points in the audio sampling signal within a preset reading window to obtain interpolation results, and then filtering out true peak detection results from the interpolation results, includes: Taking three consecutive discrete sampling points within the preset reading window as a group, parabolic interpolation is performed on the three discrete sampling points in each group to obtain the corresponding interpolation peak value for each group. The maximum value among all the interpolated peak values ​​is selected as the true peak detection result.

[0007] In one embodiment, the step of generating the current envelope signal by performing envelope following based on the true peak detection result and the release time coefficient includes: The true peak detection result is compared with the envelope signal at the previous moment; When the true peak detection result is greater than the envelope signal at the previous moment, a first-order filter update is performed based on the preset attack coefficient and the true peak detection result to obtain the envelope signal at the current moment. When the true peak detection result is not greater than the envelope signal at the previous moment, the release time coefficient and the true peak detection result are used to perform a first-order filter update to obtain the envelope signal at the current moment.

[0008] In one embodiment, the step of determining the release time coefficient based on the current signal level and a preset limiting threshold includes: Calculate the difference between the current signal level and the preset amplitude limiting threshold in the decibel domain; Calculate the adaptive weighting factor between the fast release time constant and the slow release time constant based on the difference; The release coefficient is obtained by weighting the release coefficient corresponding to the fast release time constant and the release coefficient corresponding to the slow release time constant according to the adaptive weighting factor.

[0009] In one embodiment, the step of determining the initial gain value based on the current envelope signal and the preset amplitude limiting threshold includes: Obtain a preset knee width value, and determine a lower threshold and an upper threshold based on the preset limiting threshold and the preset knee width value; Compare the current envelope signal with the lower threshold and the upper threshold; When the envelope signal at the current moment is not greater than the lower threshold, the initial gain value is determined to be 1; When the envelope signal at the current time is between the lower threshold and the upper threshold, the initial gain value of the transition region is calculated by interpolation. When the envelope signal at the current moment is not less than the upper limit threshold, the initial gain value of the compression zone is calculated according to the preset compression ratio.

[0010] In one embodiment, the step of limiting the variation range of the initial gain value to obtain the target gain value includes: Calculate the gain difference between the initial gain value of the current sampling point and the target gain value of the previous sampling point; The gain difference is limited according to a preset maximum change step size range to obtain a limited gain difference; The target gain value of the current sampling point is determined based on the limiting gain difference and the target gain value of the previous sampling point.

[0011] In one embodiment, before the step of obtaining the amplitude-limited audio signal based on the target gain value and the target sampling point signal, the method further includes: Obtain the position information of the target sampling point signal on the time axis; The target gain value and the target sampling point signal are aligned on the time axis based on the location information.

[0012] Furthermore, to achieve the above objectives, this application also proposes a digital audio true peak limiting device, the digital audio true peak limiting device comprising: The acquisition module is used to acquire audio sampling signals, perform interpolation operations on discrete sampling points in the audio sampling signals within a preset reading window to obtain interpolation results, and filter out true peak detection results from the interpolation results; The generation module is used to convert the envelope signal of the previous moment into the current signal level, determine the release time coefficient according to the current signal level and the preset amplitude limiting threshold, and perform envelope tracking according to the true peak detection result and the release time coefficient to generate the envelope signal of the current moment. The determining module is used to determine an initial gain value based on the current envelope signal and the preset amplitude limiting threshold, and to limit the change range of the initial gain value to obtain a target gain value; The limiting module is used to determine the target sampling point signal from the audio sampling signal, and to obtain a limited audio signal based on the target gain value and the target sampling point signal. The sampling point of the target sampling point signal is determined by the preset reading window.

[0013] In addition, to achieve the above objectives, this application also proposes a digital audio true peak limiting device, the device comprising: a memory, a processor, and a computer program stored in the memory and executable on the processor, the computer program being configured to implement the steps of the digital audio true peak limiting method as described above.

[0014] In addition, to achieve the above objectives, this application also proposes a storage medium, which is a computer-readable storage medium, on which a computer program is stored, and when the computer program is executed by a processor, it implements the steps of the digital audio true peak limiting method described above.

[0015] In addition, to achieve the above objectives, this application also provides a computer program product, which includes a computer program that, when executed by a processor, implements the steps of the digital audio true peak limiting method described above.

[0016] This application provides a digital audio true peak limiting method. The method involves acquiring an audio sampling signal, performing interpolation on discrete sampling points within a preset reading window to obtain interpolation results, and filtering out true peak detection results from the interpolation results. The method then converts the envelope signal from the previous moment into the current signal level, determines a release time coefficient based on the current signal level and a preset limiting threshold, and performs envelope following based on the true peak detection results and the release time coefficient to generate the current moment's envelope signal. Finally, it determines an initial gain value based on the current moment's envelope signal and the preset limiting threshold, and limits the variation of the initial gain value to obtain a target gain value. The method also determines a target sampling point signal from the audio sampling signal, and obtains a limited audio signal based on the target gain value and the target sampling point signal. The sampling points of the target sampling point signal are determined by the preset reading window. This application improves the output sound quality of the audio signal after limiting by using a pre-read window to advance the gain decision before the signal output to eliminate transient lag, estimating the true peak value in the analog domain through interpolation to avoid implicit clipping, dynamically adjusting the release time coefficient through the relationship between the signal level and the threshold to eliminate the breathing effect, and limiting the amplitude of gain value changes to avoid distortion introduced by gain abrupt changes. Attached Figure Description

[0017] The accompanying drawings, which are incorporated in and form part of this specification, illustrate embodiments consistent with this application and, together with the description, serve to explain the principles of this application.

[0018] To more clearly illustrate the technical solutions in the embodiments of this application or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, for those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0019] Figure 1 This is a flowchart illustrating an embodiment of the digital audio true peak limiting method of this application. Figure 2 This is a flowchart illustrating Embodiment 2 of the digital audio true peak limiting method of this application; Figure 3 This is a schematic diagram of the module structure of the digital audio true peak limiting device according to an embodiment of this application; Figure 4This is a schematic diagram of the device structure of the hardware operating environment involved in the digital audio true peak limiting method in the embodiments of this application.

[0020] The purpose, features, and advantages of this application will be further explained in conjunction with the embodiments and with reference to the accompanying drawings. Detailed Implementation

[0021] It should be understood that the specific embodiments described herein are merely illustrative of the technical solutions of this application and are not intended to limit this application.

[0022] To better understand the technical solution of this application, a detailed description will be provided below in conjunction with the accompanying drawings and specific implementation methods.

[0023] This application acquires an audio sampling signal, performs interpolation on discrete sampling points in the audio sampling signal within a preset reading window to obtain interpolation results, and filters out true peak detection results from the interpolation results; converts the envelope signal of the previous moment into the current signal level, determines the release time coefficient based on the current signal level and a preset limiting threshold, and performs envelope following based on the true peak detection results and the release time coefficient to generate the current moment envelope signal; determines an initial gain value based on the current moment envelope signal and the preset limiting threshold, and limits the variation amplitude of the initial gain value to obtain a target gain value; determines a target sampling point signal from the audio sampling signal, and obtains a limited audio signal based on the target gain value and the target sampling point signal, wherein the sampling points of the target sampling point signal are determined by the preset reading window.

[0024] In the field of digital audio processing, limiters are a key module for preventing signal overload. Current limiters typically rely on peak detection at sampling points, applying attenuation gain if the signal exceeds a preset threshold. However, existing solutions only detect the amplitude of discrete sampling points, failing to reflect the true peak value in the analog domain after digital-to-analog conversion, and are prone to implicit clipping. Furthermore, their gain release time is a fixed value, which cannot simultaneously address transient response and background noise stability when the signal has a large dynamic range. Therefore, improving the output sound quality of the audio signal after limiter processing remains a problem that needs to be solved.

[0025] This application improves the output sound quality of the audio signal after limiting by using a pre-read window to advance the gain decision before the signal output to eliminate transient lag, estimating the true peak value in the analog domain through interpolation to avoid implicit clipping, dynamically adjusting the release time coefficient through the relationship between the signal level and the threshold to eliminate the breathing effect, and limiting the amplitude of gain value changes to avoid distortion introduced by gain abrupt changes.

[0026] All user-related data involved in this application were obtained with the user's permission or consent; that is, when this application is applied to specific products or technologies, user permission is required to obtain and process the relevant data, and the processing of the relevant data must comply with the relevant laws, regulations and regulatory standards of the relevant countries and regions.

[0027] Based on this, embodiments of this application provide a digital audio true peak limiting method, referring to... Figure 1 , Figure 1 This is a flowchart illustrating the first embodiment of the digital audio true peak limiting method of this application.

[0028] In this embodiment, the digital audio true peak limiting method includes steps S10~S40: Step S10: Acquire the audio sampling signal, perform interpolation operation on the discrete sampling points in the audio sampling signal within a preset reading window to obtain the interpolation result, and filter out the true peak detection result from the interpolation result; It should be noted that the executing entity in this embodiment can be a computing service device with data processing, network communication, and program execution functions, such as a tablet computer, personal computer, or mobile phone, or an electronic device or digital audio true peak limiting device capable of the above functions. The following description uses a digital audio true peak limiting device as an example to illustrate this embodiment and the subsequent embodiments.

[0029] It should be noted that the audio sampling signal is a discrete-time sequence after analog-to-digital conversion, represented as x[n], where n is the discrete-time index, representing the sequence number of the sampling point. After acquiring the audio sampling signal, the system can process the signal within a preset reading window. The preset reading window is a predefined data range, for example, from the current sampling point n to the next L sampling points n+L. In the specific implementation, the input audio signal x[n] is not processed directly, but is first written to a circular buffer. The circular buffer is a first-in-first-out circular storage structure that can achieve delay and pre-fetching functions without moving data. Then, the signal of the current sampling point and the next L sampling points after the current sampling point, i.e., x[n] to x[n+L], is read from the circular buffer for gain decision. At the same time, the signal x[n - L], which is delayed by L sampling points, can be read as the current output signal, i.e.:

[0030] in, The output signal is defined as L, which is the pre-read length, specifically 128 to 512 sampling points, corresponding to a pre-read time of approximately 2.6 milliseconds (ms) to 10 ms at a sampling rate of 48 kHz. Within the preset read window, the system performs interpolation on the discrete sampling points. Due to the reconstruction filtering during the digital-to-analog conversion process, the peak value of the analog waveform between sampling points is often higher than the peak value of the digital sampling. Traditional detection based on the peak value of sampling points cannot reflect this true peak value. Therefore, this embodiment performs oversampling interpolation on the signal within the window to estimate the peak value of the analog waveform, so that the detection result can reflect the true peak value of the analog domain after digital-to-analog conversion. After the interpolation operation is completed, the interpolation result is obtained, which includes the peak value estimated by interpolation for each group of sampling points within the window. Finally, the maximum value can be selected from all interpolation results as the true peak detection result, denoted as d[n].

[0031] In one feasible approach, the step of performing interpolation on discrete sampling points in the audio sampling signal within a preset reading window to obtain interpolation results, and then selecting the true peak detection result from the interpolation results, includes: taking three consecutive discrete sampling points within the preset reading window as a group, performing parabolic interpolation on the three discrete sampling points in each group to obtain the corresponding interpolated peak value for each group; and selecting the maximum value from all the interpolated peak values ​​as the true peak detection result.

[0032] It should be noted that the signal within the pre-read window is sampled and interpolated to estimate the peak value of the analog waveform. This embodiment uses the computationally efficient parabolic interpolation method.

[0033] Specifically, for three consecutive sampling points x[k - 1], x[k], x[k + 1], calculate the peak offset p and the interpolated peak value. :

[0034]

[0035] The detector outputs d[n], where d[n] is the maximum value of all interpolated peaks within the prefetch window.

[0036] in, This is the true peak detection result.

[0037] Step S20: Convert the envelope signal of the previous moment into the current signal level, determine the release time coefficient based on the current signal level and the preset amplitude limiting threshold, and perform envelope tracking based on the true peak detection result and the release time coefficient to generate the envelope signal of the current moment; It should be noted that the system first converts the envelope signal e[n-1] from the previous moment into the current signal level. The envelope signal is the contour curve of the audio signal amplitude changing over time, reflecting the macroscopic trend of sound loudness. The envelope signal e[n-1] from the previous moment is the envelope value that has been calculated and stored at the previous sampling point. In the digital signal processing implementation, the system needs to initialize the envelope history state e[-1] to 0 when it starts up. When the first valid sampling point arrives, the system enters the attack state and generates the initial envelope according to the attack coefficient. Converting e[n-1] into the current signal level is essentially a decibel (dB) domain conversion of the envelope value in the linear domain. The resulting decibel domain value is the current signal level. Specifically, the envelope signal e[n-1] from the previous moment is the linear amplitude value of the envelope signal that has been calculated and stored at the previous sampling point, with a value range of 0 to 1.0 (corresponding to full digital amplitude). The conversion formula for converting this linear amplitude value into the decibel domain value is as follows:

[0038] in, This is the decibel threshold, i.e., the current signal level. () is the standard formula for converting linear amplitude to decibel value; e[n-1] is the envelope signal of the previous time step.

[0039] Then, the system determines the release time coefficient based on the current signal level and a preset limiting threshold. The preset limiting threshold is a system-defined amplitude limit, representing the maximum amplitude boundary allowed by the limiter. This threshold serves as a core reference threshold in both adaptive release and gain calculation. Release Time Coefficient It is a time-varying parameter. The system dynamically adjusts the signal based on the current signal level. Then, it uses the true peak detection result d[n] and the release time coefficient... Envelope following is performed to generate the current envelope signal e[n]. Envelope following is a point-by-point recursive process, where the current envelope value is determined by the current peak detection result and the envelope value of the previous moment. The system can compare the true peak detection result with the envelope signal of the previous moment to determine whether it is currently in the attack phase or the release phase, and thus adopt different update coefficients. Through this process, the system can generate the current envelope signal e[n].

[0040] In one feasible approach, the step of determining the release time coefficient based on the current signal level and the preset amplitude limiting threshold includes: calculating the difference between the current signal level and the preset amplitude limiting threshold in the decibel domain; calculating an adaptive weighting factor between the fast release time constant and the slow release time constant based on the difference; and weighting the release coefficient corresponding to the fast release time constant and the release coefficient corresponding to the slow release time constant based on the adaptive weighting factor to obtain the release time coefficient.

[0041] It should be noted that the preset limiting threshold, i.e., the threshold of the limiter, is... .

[0042] For the preset amplitude limiting threshold, first convert it to a decibel range value:

[0043] in, This is the decibel range of the preset amplitude limiting threshold.

[0044] The difference (dB range) between the current level and the preset limiting threshold is: =

[0045] in, Indicates the difference. This is the decibel threshold.

[0046] It should be noted that a fast release time constant is used when the signal significantly exceeds the threshold and the gain needs to be restored as quickly as possible to avoid loudness loss or distortion caused by prolonged suppression. A slow release time constant is used when the signal falls back to near or below the threshold to slowly restore the gain, preventing background noise (floor noise) from producing audible breathing sounds or pumping effects as the gain rapidly recovers.

[0047] Adaptive weighting factor The Sigmoid function is used for calculation, and the specific formula is as follows:

[0048] Where m represents the transition steepness, a preset constant that determines the transition rate of the Sigmoid function near the switching point. A larger m value results in a steeper transition near the switching point, while a smaller m value results in a smoother transition. This value is preset based on actual listening tests and system requirements. The switching threshold is a preset decibel value that determines the center position of the fast / slow release switching.

[0049] The system is pre-designed with a fast release time constant. and slow release time constant Fast release time constant Used for scenarios requiring rapid gain recovery when the signal is much higher than the threshold, for example, set to 50ms; slow release time constant. This is used for slow gain recovery scenarios when the signal is below a threshold, for example, set to 500ms. The two time constants correspond to two speed presets for the envelope follower release phase. The envelope smoothing coefficients corresponding to the two time constants are expressed as follows: and The envelope update formula for a first-order filter is:

[0050]

[0051] in, The system determines the sampling rate based on the sampling rate and a preset slow release time constant. Calculate the release coefficient corresponding to the slow release time constant. Based on the sampling rate and the preset slow release time constant Calculate the release coefficient corresponding to the slow release time constant. .

[0052] Adaptive weighting factor Used in and Interpolation is performed between these values ​​to obtain the final release coefficient used in actual applications. :

[0053] Step S30: Determine the initial gain value based on the current envelope signal and the preset amplitude limiting threshold, and limit the change range of the initial gain value to obtain the target gain value; It should be noted that the purpose of this step is to calculate the gain value required for amplitude limiting and to smooth the gain value. First, the system determines the initial gain value based on the current envelope signal e[n] and the preset amplitude limiting threshold. The initial gain value is the theoretically desired instantaneous gain multiple, and its value does not exceed 1. When the envelope signal is much lower than the threshold, the signal does not need compression, the gain is 1, indicating that the signal is direct; when the envelope signal is close to the threshold, the gain smoothly transitions between 1 and the compression value; when the envelope signal is much higher than the threshold, the gain is attenuated according to the preset compression ratio. Then, the system limits the change amplitude of the initial gain value to obtain the target gain value. Even if the initial gain value itself is smooth, if the gain value changes too quickly between adjacent sampling points, it will introduce audible modulation noise. Therefore, the system can limit the rate of gain change to ensure that the gain change at each sampling point does not exceed the preset maximum step size, thereby filtering out the high-frequency components of the gain change. The output gain value after limiting the change amplitude is the final target gain value actually applied to the signal.

[0054] In one feasible approach, the step of determining the initial gain value based on the current envelope signal and the preset amplitude limiting threshold includes: obtaining a preset knee width value, and determining a lower threshold and an upper threshold based on the preset amplitude limiting threshold and the preset knee width value; comparing the current envelope signal with the lower threshold and the upper threshold; determining the initial gain value as 1 when the current envelope signal is not greater than the lower threshold; calculating the initial gain value of the transition zone by interpolation when the current envelope signal is between the lower threshold and the upper threshold; and calculating the initial gain value of the compression zone based on a preset compression ratio when the current envelope signal is not less than the upper threshold.

[0055] It should be noted that the gain value can be calculated using a soft-knee curve. The soft-knee curve is used to avoid the distortion caused by hard switching from unrestricted to restricted near the threshold, making the gain change smoother and avoiding hard switching distortion.

[0056] The current envelope signal is The preset limit threshold is The compression ratio is R, and the knee width is W.

[0057] All values ​​are calculated in the linear domain, and the calculation formula is as follows:

[0058] in, The initial gain value is represented by the compression ratio R, which represents the strength of gain attenuation after exceeding the threshold. It is defined as the output level increasing by 1 / R dB for every 1 dB increase in input level. The lower threshold is... This is the upper limit threshold.

[0059] The knee width W defines the amplitude range of the smooth transition near the threshold, making the transition curve from uncompressed to fully compressed gain continuously differentiable. Typical values ​​are between 3 dB and 10 dB. For example, W=6 dB indicates a transition zone from 3 dB below the threshold to 3 dB above it.

[0060] When the signal is much lower than the threshold ( ): No compression, gain = 1, signal pass-through.

[0061] When the signal is much higher than the threshold ( ): Entering the hard-limit region, the gain decreases linearly with the compression ratio, limiting the output amplitude to . .

[0062] Within the knee width range: the gain is interpolated between 1 and the hard limit value, typically using a quadratic or linear interpolation method to ensure the gain curve remains continuously differentiable and avoids abrupt distortion. Here... It is the desired instantaneous gain factor (≤1).

[0063] In one feasible approach, the step of limiting the variation range of the initial gain value to obtain the target gain value includes: calculating the gain difference between the initial gain value of the current sampling point and the target gain value of the previous sampling point; limiting the gain difference according to a preset maximum change step size range to obtain a limited gain difference; and determining the target gain value of the current sampling point based on the limited gain difference and the target gain value of the previous sampling point.

[0064] It should be noted that even if the gain curve itself is smooth, if the gain value changes too rapidly between adjacent sampling points, audible modulation noise will be introduced. Therefore, it is necessary to limit the rate of change of the gain, i.e., slope limiting.

[0065] First, calculate the initial gain value. Gain compared to the previous sampling point The difference:

[0066] Then, limit this difference to the maximum allowable step size. Within, that is:

[0067] Here, clamp(x, min, max) means truncating x within the interval [min, max].

[0068] The equivalent representation is:

[0069] in, The purpose of clamp is to ensure that the gain does not exceed the target gain value in each change. This filters out the high-frequency components of gain changes, thus avoiding the introduction of new distortions due to rapid gain jumps. The parameter can be set according to the sampling rate and acceptable setup time, with typical values ​​corresponding to a variation of approximately 0.001 to 0.01 dB per sampling point. The final output gain g[n] follows both a soft-knee curve and has smooth time evolution characteristics.

[0070] Step S40: Determine the target sampling point signal from the audio sampling signal, and obtain a limited audio signal based on the target gain value and the target sampling point signal. The sampling point of the target sampling point signal is determined by the preset reading window.

[0071] It should be noted that the calculated target gain value can be applied to the audio signal to complete the limiting process. First, the system determines the target sampling point signal from the audio sample signal. The target sampling point signal is a signal delayed by a preset number of sampling points L relative to the current input sampling point, denoted as x[nL]. In the specific implementation, the system reads the signal x[nL] delayed by L sampling points from the circular buffer as the current output signal. This signal is the target sampling point signal that lags behind the current input sampling point in time. The lag of the target sampling point signal is determined by the length L of the preset reading window. The preset reading window looks forward by L points (x[n]...x[n+L]), while the target sampling point signal looks backward by L points (x[nL]). Both use the same value L to ensure that the signal window used for gain decision and the signal on which the gain is applied have a definite temporal correspondence. Then, the system obtains the limited audio signal y[n] based on the target gain value g[n] and the target sampling point signal x[nL]. Specifically, the system multiplies the calculated gain g[n] by the delayed original signal x[nL], i.e.:

[0072] Therefore, the target gain is applied to the target sampling point signal. Signal amplitudes exceeding the threshold are attenuated to below the threshold, while signals below the threshold are allowed to pass as is. The resulting y[n] is the amplitude-limited audio signal, which is then output to the next level audio link through the input / output interface. Since the gain is calculated based on a preset reading window (including current and future sampling points) but is applied to the delayed signal (x[nL]), the gain adjustment precedes the arrival of the signal peak, achieving causal compensation.

[0073] In one feasible approach, before the step of obtaining the amplitude-limited audio signal based on the target gain value and the target sampling point signal, the method further includes: acquiring the position information of the target sampling point signal on the time axis; and aligning the target gain value and the target sampling point signal on the time axis based on the position information.

[0074] It should be noted that the system also acquires the position information of the target sampling point signal x[nL] on the time axis. In the circular buffer implementation, each sampling point carries its corresponding time index or position mark when written to the buffer. The target sampling point signal x[nL] is the signal of L sampling points before the current input sampling point x[n], and it is located on the time axis at a position L sampling periods before the current moment. Based on the acquired position information, the system aligns the target gain value g[n] with the target sampling point signal x[nL] on the time axis. Specifically, the gain value g[n] is calculated based on a preset reading window (including x[n]...x[n+L]), and it needs to be applied to the delayed signal x[nL]. To ensure a strict correspondence between the gain and the signal on the time axis, the system can align the calculated gain value g[n] with the delayed signal x[nL] in time, that is, the gain value g[n] acts on the signal x[nL], rather than on the current input signal x[n] or the future signal. Once alignment is complete, the system can apply the target gain value to the target sampling point signal to complete the amplitude limiting process.

[0075] This embodiment acquires an audio sampling signal, performs interpolation on discrete sampling points in the audio sampling signal within a preset reading window to obtain interpolation results, and filters out true peak detection results from the interpolation results; converts the envelope signal of the previous moment into the current signal level, determines the release time coefficient based on the current signal level and a preset limiting threshold, and performs envelope following based on the true peak detection results and the release time coefficient to generate the current moment envelope signal; determines an initial gain value based on the current moment envelope signal and the preset limiting threshold, and limits the variation amplitude of the initial gain value to obtain a target gain value; determines a target sampling point signal from the audio sampling signal, and obtains a limited audio signal based on the target gain value and the target sampling point signal, wherein the sampling points of the target sampling point signal are determined by the preset reading window. This embodiment improves the output sound quality of the audio signal after limiting by using a pre-read window to advance the gain decision before the signal output to eliminate transient lag, estimating the true peak value in the analog domain through interpolation to avoid implicit clipping, dynamically adjusting the release time coefficient based on the relationship between the signal level and the threshold to eliminate the breathing effect, and limiting the amplitude of gain value changes to avoid distortion introduced by sudden gain changes.

[0076] Based on the first embodiment of this application, in the second embodiment of this application, the content that is the same as or similar to that in Embodiment 1 above can be referred to the above description, and will not be repeated hereafter. Based on this, please refer to... Figure 2 Step S20 also includes steps S201 to S203: Step S201: Compare the true peak detection result with the envelope signal from the previous moment; It should be noted that the system compares the true peak detection result d[n] at the current sampling point with the envelope signal e[n-1] at the previous sampling point. This comparison determines whether the envelope follower is currently in an attack state or a release state. The attack state and release state are used to describe the direction of the envelope signal's response to changes in the amplitude of the input signal. When the signal amplitude is increasing, the envelope follower enters the attack state; when the signal amplitude is decreasing or remains stable, the envelope follower enters the release state.

[0077] In the implementation of digital signal processing, when the system starts, the envelope history state e[-1] can be initialized to 0. When the first valid sampling point arrives, since d[0]>0 and d[0]>e[-1] (e[-1]=0), the system enters the attack state and generates the initial envelope according to the attack coefficient.

[0078] Step S202: When the true peak detection result is greater than the envelope signal at the previous moment, perform first-order filtering update based on the preset attack coefficient and the true peak detection result to obtain the envelope signal at the current moment; It should be noted that the envelope signal e[n] can be generated based on the true peak detection result, and the envelope update formula for the attack state is:

[0079] Where attack represents the update formula for the attack state, and release represents the update formula for the release state. This indicates the true peak detection result. This represents the preset attack coefficient, which is a fixed constant and corresponds to an extremely fast attack time. This represents the release time coefficient.

[0080] Step S203: When the true peak detection result is not greater than the envelope signal of the previous moment, the release time coefficient and the true peak detection result are used to perform first-order filtering and update to obtain the envelope signal of the current moment.

[0081] It should be noted that when d[n] ≤ e[n-1], it indicates that the signal amplitude is decreasing or remaining constant, and the system is in the release phase. At this time, the envelope update formula is:

[0082] Where, release represents the update formula for the release state. This indicates the true peak detection result. This represents the release time coefficient.

[0083] This embodiment compares the true peak detection result with the envelope signal from the previous moment. When the true peak detection result is greater than the envelope signal from the previous moment, a first-order filter update is performed based on a preset attack coefficient and the true peak detection result to obtain the current envelope signal. When the true peak detection result is not greater than the envelope signal from the previous moment, a first-order filter update is performed using the release time coefficient and the true peak detection result to obtain the current envelope signal. By comparing the true peak detection result with the envelope signal from the previous moment, this embodiment enables the envelope follower to respond quickly with a fixed attack coefficient when the signal rises, ensuring that transient peaks are not missed. When the signal falls, a dynamically determined release time coefficient is used for smooth fallback, avoiding auditory distortion introduced by gain abrupt changes. This achieves a balance between rapid tracking and smooth transition of the envelope signal to signal amplitude changes.

[0084] It should be noted that the above examples are only for understanding this application and do not constitute a limitation on the digital audio true peak limiting method of this application. Any simple modifications based on this technical concept are within the protection scope of this application.

[0085] This application also provides a digital audio true peak limiting device; please refer to... Figure 3 The digital audio true peak limiting device includes: The acquisition module 10 is used to acquire audio sampling signals, perform interpolation operations on discrete sampling points in the audio sampling signals within a preset reading window to obtain interpolation results, and filter out true peak detection results from the interpolation results; The generation module 20 is used to convert the envelope signal of the previous moment into the current signal level, determine the release time coefficient according to the current signal level and the preset amplitude limiting threshold, and perform envelope tracking according to the true peak detection result and the release time coefficient to generate the envelope signal of the current moment. The determining module 30 is used to determine an initial gain value based on the current envelope signal and the preset amplitude limiting threshold, and to limit the change range of the initial gain value to obtain a target gain value; The limiting module 40 is used to determine the target sampling point signal from the audio sampling signal, and to obtain a limited audio signal based on the target gain value and the target sampling point signal. The sampling point of the target sampling point signal is determined by the preset reading window.

[0086] This application acquires an audio sampling signal, performs interpolation on discrete sampling points in the audio sampling signal within a preset reading window to obtain interpolation results, and filters out true peak detection results from the interpolation results; converts the envelope signal of the previous moment into the current signal level, determines the release time coefficient based on the current signal level and a preset limiting threshold, and performs envelope following based on the true peak detection results and the release time coefficient to generate the current moment envelope signal; determines an initial gain value based on the current moment envelope signal and the preset limiting threshold, and limits the variation amplitude of the initial gain value to obtain a target gain value; determines a target sampling point signal from the audio sampling signal, and obtains a limited audio signal based on the target gain value and the target sampling point signal, wherein the sampling points of the target sampling point signal are determined by the preset reading window. This application improves the output sound quality of the audio signal after limiting by using a pre-read window to advance the gain decision before the signal output to eliminate transient lag, estimating the true peak value in the analog domain through interpolation to avoid implicit clipping, dynamically adjusting the release time coefficient through the relationship between the signal level and the threshold to eliminate the breathing effect, and limiting the amplitude of gain value changes to avoid distortion introduced by gain abrupt changes.

[0087] In one embodiment, the acquisition module 10 is further configured to take three consecutive discrete sampling points within the preset reading window as a group, perform parabolic interpolation on the three discrete sampling points in each group to obtain the corresponding interpolation peak value for each group; and select the maximum value from all the interpolation peak values ​​as the true peak detection result.

[0088] In one embodiment, the generation module 20 is further configured to compare the true peak detection result with the envelope signal at the previous moment; when the true peak detection result is greater than the envelope signal at the previous moment, a first-order filter update is performed based on a preset attack coefficient and the true peak detection result to obtain the envelope signal at the current moment; when the true peak detection result is not greater than the envelope signal at the previous moment, a first-order filter update is performed based on the release time coefficient and the true peak detection result to obtain the envelope signal at the current moment.

[0089] In one embodiment, the generation module 20 is further configured to calculate the difference between the current signal level and the preset amplitude limiting threshold in the decibel domain; calculate an adaptive weighting factor between the fast release time constant and the slow release time constant based on the difference; and perform weighting processing on the release coefficient corresponding to the fast release time constant and the release coefficient corresponding to the slow release time constant based on the adaptive weighting factor to obtain the release time coefficient.

[0090] In one embodiment, the determining module 30 is further configured to obtain a preset knee width value, and determine a lower threshold and an upper threshold based on the preset amplitude limiting threshold and the preset knee width value; compare the current envelope signal with the lower threshold and the upper threshold; when the current envelope signal is not greater than the lower threshold, determine the initial gain value as 1; when the current envelope signal is between the lower threshold and the upper threshold, calculate the initial gain value of the transition region by interpolation; when the current envelope signal is not less than the upper threshold, calculate the initial gain value of the compression region based on a preset compression ratio.

[0091] In one embodiment, the determining module 30 is further configured to calculate the gain difference between the initial gain value of the current sampling point and the target gain value of the previous sampling point; limit the gain difference according to a preset maximum change step size range to obtain a limited gain difference; and determine the target gain value of the current sampling point according to the limited gain difference and the target gain value of the previous sampling point.

[0092] In one embodiment, the limiting module 40 is further configured to acquire the position information of the target sampling point signal on the time axis; and to align the target gain value with the target sampling point signal on the time axis according to the position information.

[0093] The digital audio true peak limiting device provided in this application, employing the digital audio true peak limiting method described in the above embodiments, can solve the technical problem of how to improve the output sound quality of the audio signal after limiting processing. Compared with the prior art, the beneficial effects of the digital audio true peak limiting device provided in this application are the same as those of the digital audio true peak limiting method described in the above embodiments, and other technical features in the digital audio true peak limiting device are the same as those disclosed in the methods of the above embodiments, and will not be repeated here.

[0094] This application provides a digital audio true peak limiting device, which includes: at least one processor; and a memory communicatively connected to the at least one processor; wherein the memory stores instructions executable by the at least one processor, which are executed by the at least one processor to enable the at least one processor to perform the digital audio true peak limiting method in the first embodiment described above.

[0095] The following is for reference. Figure 4This document illustrates a structural schematic diagram of a digital audio true peak limiting device suitable for implementing embodiments of this application. The digital audio true peak limiting device in embodiments of this application may include, but is not limited to, mobile terminals such as mobile phones, laptops, digital broadcast receivers, PDAs (Personal Digital Assistants), tablets, PMPs (Portable Media Players), in-vehicle terminals (e.g., in-vehicle navigation terminals), and fixed terminals such as digital TVs and desktop computers. Figure 4 The digital audio true peak limiting device shown is merely an example and should not impose any limitation on the functionality and scope of use of the embodiments of this application.

[0096] like Figure 4 As shown, the digital audio true peak limiting device may include a processing unit 1001 (e.g., a central processing unit, a graphics processing unit, etc.) that can perform various appropriate actions and processes according to a program stored in ROM (Read Only Memory) 1002 or a program loaded from storage device 1003 into RAM (Random Access Memory) 1004. RAM 1004 also stores various programs and data required for the operation of the digital audio true peak limiting device. The processing unit 1001, ROM 1002, and RAM 1004 are interconnected via bus 1005. Input / output (I / O) interface 1006 is also connected to the bus. Typically, the following systems can be connected to I / O interface 1006: input devices 1007 including, for example, touchscreens, touchpads, keyboards, mice, image sensors, microphones, accelerometers, gyroscopes, etc.; output devices 1008 including, for example, liquid crystal displays (LCDs), speakers, vibrators, etc.; storage devices 1003 including, for example, magnetic tapes, hard disks, etc.; and communication devices 1009. Communication device 1009 allows the digital audio true peak limiter to communicate wirelessly or wiredly with other devices to exchange data. Although digital audio true peak limiters with various systems are shown in the figures, it should be understood that it is not required to implement or possess all of the systems shown. More or fewer systems may be implemented alternatively.

[0097] Specifically, according to the embodiments disclosed in this application, the processes described above with reference to the flowcharts can be implemented as computer software programs. For example, embodiments disclosed in this application include a computer program product comprising a computer program carried on a computer-readable medium, the computer program containing program code for performing the methods shown in the flowcharts. In such embodiments, the computer program can be downloaded and installed from a network via a communication device, or installed from storage device 1003, or installed from ROM 1002. When the computer program is executed by processing device 1001, it performs the functions defined in the methods of the embodiments disclosed in this application.

[0098] The digital audio true peak limiting device provided in this application, employing the digital audio true peak limiting method described in the above embodiments, can solve the technical problem of how to improve the output sound quality of the audio signal after limiting processing. Compared with the prior art, the beneficial effects of the digital audio true peak limiting device provided in this application are the same as those of the digital audio true peak limiting method provided in the above embodiments, and other technical features of this digital audio true peak limiting device are the same as those disclosed in the previous embodiment method, and will not be repeated here.

[0099] It should be understood that the various parts disclosed in this application can be implemented using 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 suitable manner in one or more embodiments or examples.

[0100] The above description is merely a specific embodiment of this application, but the scope of protection of this application is not limited thereto. Any variations or substitutions that can be easily conceived by those skilled in the art within the scope of the technology disclosed in this application should be included within the scope of protection of this application. Therefore, the scope of protection of this application should be determined by the scope of the claims.

[0101] This application provides a computer-readable storage medium having computer-readable program instructions (i.e., a computer program) stored thereon, the computer-readable program instructions being used to execute the digital audio true peak limiting method in the above embodiments.

[0102] The computer-readable storage medium provided in this application may be, for example, a USB flash drive, but is not limited to, electrical, magnetic, optical, electromagnetic, infrared, or semiconductor systems, devices, or any combination thereof. More specific examples of computer-readable storage media may include, but are not limited to: electrical connections having one or more wires, portable computer disks, hard disks, random access memory (RAM), read-only memory (ROM), erasable programmable read-only memory (EPROM or flash memory), optical fiber, portable compact disk read-only memory (CD-ROM), optical storage devices, magnetic storage devices, or any suitable combination thereof. 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 conjunction with an instruction execution system, system, or device. The program code contained on the computer-readable storage medium may be transmitted using any suitable medium, including but not limited to: wires, optical cables, RF (Radio Frequency), etc., or any suitable combination thereof.

[0103] The aforementioned computer-readable storage medium may be included in a digital audio true peak limiter; or it may exist independently and not assembled into a digital audio true peak limiter.

[0104] The aforementioned computer-readable storage medium carries one or more programs. When these programs are executed by the digital audio true peak limiting device, the digital audio true peak limiting device: acquires an audio sampling signal; performs interpolation operations on discrete sampling points in the audio sampling signal within a preset reading window to obtain an interpolation result; and filters out a true peak detection result from the interpolation result; converts the envelope signal from the previous moment into a current signal level; determines a release time coefficient based on the current signal level and a preset limiting threshold; performs envelope following based on the true peak detection result and the release time coefficient to generate a current moment envelope signal; determines an initial gain value based on the current moment envelope signal and the preset limiting threshold; and limits the variation amplitude of the initial gain value to obtain a target gain value; determines a target sampling point signal from the audio sampling signal; and obtains a limited audio signal based on the target gain value and the target sampling point signal, wherein the sampling point of the target sampling point signal is determined by the preset reading window.

[0105] Computer program code for performing the operations of this application can be written in one or more programming languages ​​or a combination thereof, including object-oriented programming languages ​​such as Java, Smalltalk, and C++, and conventional procedural programming languages ​​such as the "C" language or similar programming languages. The program code can be executed entirely on the user's computer, partially on the user's computer, as a standalone software package, partially on the user's computer and partially on a remote computer, or entirely on a remote computer or server. In cases involving remote computers, the remote computer can be connected to the user's computer via any type of network—including a Local Area Network (LAN) or a Wide Area Network (WAN)—or can be connected to an external computer (e.g., via the Internet using an Internet service provider).

[0106] The flowcharts and block diagrams in the accompanying drawings illustrate the architecture, functionality, and operation of possible implementations of systems, methods, and computer program products according to various embodiments of this application. In this regard, each block in a flowchart or block diagram may represent a module, segment, or portion of code containing one or more executable instructions for implementing a specified logical function. It should also be noted that in some alternative implementations, the functions indicated in the blocks may occur in a different order than those indicated in the drawings. For example, two consecutively indicated blocks may actually be executed substantially in parallel, and they may sometimes be executed in reverse order, depending on the functions involved. It should also be noted that each block in the block diagrams and / or flowcharts, and combinations of blocks in the block diagrams and / or flowcharts, can be implemented using a dedicated hardware-based system that performs the specified function or operation, or using a combination of dedicated hardware and computer instructions.

[0107] The modules described in the embodiments of this application can be implemented in software or hardware. The names of the modules do not necessarily limit the functionality of the unit itself.

[0108] The readable storage medium provided in this application is a computer-readable storage medium that stores computer-readable program instructions (i.e., a computer program) for executing the above-described digital audio true peak limiting method, thereby solving the technical problem of how to improve the output sound quality of the audio signal after limiting processing. Compared with the prior art, the beneficial effects of the computer-readable storage medium provided in this application are the same as those of the digital audio true peak limiting method provided in the above embodiments, and will not be repeated here.

[0109] This application also provides a computer program product, including a computer program that, when executed by a processor, implements the steps of the digital audio true peak limiting method described above.

[0110] The computer program product provided in this application can solve the technical problem of how to improve the output sound quality of audio signals after clipping processing. Compared with the prior art, the beneficial effects of the computer program product provided in this application are the same as those of the digital audio true peak clipping method provided in the above embodiments, and will not be repeated here.

[0111] The above description is only a part of the embodiments of this application and does not limit the patent scope of this application. All equivalent structural transformations made under the technical concept of this application and using the contents of the specification and drawings of this application, or direct / indirect applications in other related technical fields, are included in the patent protection scope of this application.

Claims

1. A method for true peak limiting of digital audio, characterized in that, The method includes: Acquire an audio sampling signal, perform interpolation operations on discrete sampling points in the audio sampling signal within a preset reading window to obtain interpolation results, and filter out true peak detection results from the interpolation results; The envelope signal from the previous moment is converted into the current signal level. The release time coefficient is determined based on the current signal level and a preset amplitude limiting threshold. Envelope tracking is performed based on the true peak detection result and the release time coefficient to generate the envelope signal at the current moment. An initial gain value is determined based on the current envelope signal and the preset amplitude limiting threshold, and the variation range of the initial gain value is limited to obtain the target gain value. The target sampling point signal is determined from the audio sampling signal, and a limited audio signal is obtained based on the target gain value and the target sampling point signal. The sampling point of the target sampling point signal is determined by the preset reading window.

2. The method as described in claim 1, characterized in that, The step of performing interpolation operations on discrete sampling points in the audio sampling signal within a preset reading window to obtain interpolation results, and then filtering out true peak detection results from the interpolation results includes: Taking three consecutive discrete sampling points within the preset reading window as a group, parabolic interpolation is performed on the three discrete sampling points in each group to obtain the corresponding interpolation peak value for each group. The maximum value among all the interpolated peak values ​​is selected as the true peak detection result.

3. The method as described in claim 1, characterized in that, The step of generating the current envelope signal by performing envelope following based on the true peak detection result and the release time coefficient includes: The true peak detection result is compared with the envelope signal at the previous moment; When the true peak detection result is greater than the envelope signal at the previous moment, a first-order filter update is performed based on the preset attack coefficient and the true peak detection result to obtain the envelope signal at the current moment. When the true peak detection result is not greater than the envelope signal at the previous moment, the release time coefficient and the true peak detection result are used to perform a first-order filter update to obtain the envelope signal at the current moment.

4. The method as described in claim 1, characterized in that, The step of determining the release time coefficient based on the current signal level and the preset amplitude limiting threshold includes: Convert the current signal level to a decibel range value; Calculate the difference between the current signal level and the preset amplitude limiting threshold in the decibel domain; Calculate the adaptive weighting factor between the fast release time constant and the slow release time constant based on the difference; The release coefficient is obtained by weighting the release coefficient corresponding to the fast release time constant and the release coefficient corresponding to the slow release time constant according to the adaptive weighting factor.

5. The method as described in claim 1, characterized in that, The step of determining the initial gain value based on the current envelope signal and the preset amplitude limiting threshold includes: Obtain a preset knee width value, and determine a lower threshold and an upper threshold based on the preset limiting threshold and the preset knee width value; Compare the current envelope signal with the lower threshold and the upper threshold; When the envelope signal at the current moment is not greater than the lower threshold, the initial gain value is determined to be 1; When the envelope signal at the current time is between the lower threshold and the upper threshold, the initial gain value of the transition region is calculated by interpolation. When the envelope signal at the current moment is not less than the upper limit threshold, the initial gain value of the compression zone is calculated according to the preset compression ratio.

6. The method as described in claim 1, characterized in that, The step of limiting the variation range of the initial gain value to obtain the target gain value includes: Calculate the gain difference between the initial gain value of the current sampling point and the target gain value of the previous sampling point; The gain difference is limited according to a preset maximum change step size range to obtain a limited gain difference; The target gain value of the current sampling point is determined based on the limiting gain difference and the target gain value of the previous sampling point.

7. The method as described in claim 1, characterized in that, Before the step of obtaining the amplitude-limited audio signal based on the target gain value and the target sampling point signal, the method further includes: Obtain the position information of the target sampling point signal on the time axis; The target gain value and the target sampling point signal are aligned on the time axis based on the location information.

8. A digital audio true peak limiting device, characterized in that, The device includes: The acquisition module is used to acquire audio sampling signals, perform interpolation operations on discrete sampling points in the audio sampling signals within a preset reading window to obtain interpolation results, and filter out true peak detection results from the interpolation results; The generation module is used to convert the envelope signal of the previous moment into the current signal level, determine the release time coefficient according to the current signal level and the preset amplitude limiting threshold, and perform envelope tracking according to the true peak detection result and the release time coefficient to generate the envelope signal of the current moment. The determining module is used to determine an initial gain value based on the current envelope signal and the preset amplitude limiting threshold, and to limit the change range of the initial gain value to obtain a target gain value; The limiting module is used to determine the target sampling point signal from the audio sampling signal, and to obtain a limited audio signal based on the target gain value and the target sampling point signal. The sampling point of the target sampling point signal is determined by the preset reading window.

9. A digital audio true peak limiting device, characterized in that, The device includes: a memory, a processor, and a computer program stored in the memory and executable on the processor, the computer program being configured to implement the steps of the digital audio true peak limiting method as claimed in any one of claims 1 to 7.

10. A storage medium, characterized in that, The storage medium is a computer-readable storage medium, and a computer program is stored on the storage medium. When the computer program is executed by a processor, it implements the digital audio true peak limiting method as described in any one of claims 1 to 7.

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