Audio signal processing method and device

Abstract

The embodiment of the invention provides an audio signal processing method and device, and relates to the technical field of audio signal processing. The method comprises the following steps: acquiring a first audio signal, wherein the first audio signal is an audio signal corresponding to a first sampling point; acquiring a delay signal corresponding to the first audio signal; performing delay processing on a reverberation signal corresponding to the second audio signal to obtain a delay feedback signal corresponding to the first sampling point; the second audio signal is an audio signal corresponding to a second sampling point, and the second sampling point is a sampling point before the first sampling point; and generating a reverberation signal corresponding to the first audio signal according to the delay signal corresponding to the first audio signal and the delay feedback signal corresponding to the first audio signal. The embodiment of the invention is used for improving the reverberation sound effect of sound.

Description

Technical Field

[0001] The present application relates to the technical field of audio signal processing, and in particular to a method and device for processing an audio signal. Background Art

[0002] Reverberation refers to the auditory sensation produced by sound after it has been reflected, refracted, scattered and attenuated multiple times in space before it reaches the human ear. After the sound reaches the ear directly, the residual sound formed by multiple reflections, refractions, scattering and attenuation will enhance the sense of space, sound depth and clarity in the subjective hearing of the human ear. Therefore, reverberators are widely used in music production, video production, film editing and other fields.

[0003] At present, reverberators are generally based on digital filter reverberation algorithms, which use digital filters to achieve reverberation effects by simulating the propagation and reflection process of sound waves in space. However, due to the complexity of modeling and computing resources, reverberation algorithms based on digital filters usually only consider a small number of reflections when simulating the reflection and attenuation process of sound in space, which leads to inaccurate reverberation simulation, and thus a large gap between the reverberation sound effect and the real scene. Summary of the invention

[0004] In view of this, an embodiment of the present application provides a method and device for processing an audio signal, which is used to enhance the reverberation sound effect of a sound.

[0005] In order to achieve the above purpose, the embodiments of the present application provide the following technical solutions:

[0006] In a first aspect, an embodiment of the present application provides a method for processing an audio signal, comprising:

[0007] Acquire a first audio signal, where the first audio signal is an audio signal corresponding to a first sampling point;

[0008] Acquire a delayed signal corresponding to the first audio signal;

[0009] Delaying the reverberation signal corresponding to the second audio signal to obtain a delayed feedback signal corresponding to the first sampling point; the second audio signal is an audio signal corresponding to the second sampling point, and the second sampling point is a sampling point before the first sampling point;

[0010] A reverberation signal corresponding to the first audio signal is generated according to a delayed signal corresponding to the first audio signal and a delayed feedback signal corresponding to the first audio signal.

[0011] As an optional implementation of the embodiment of the present application, the first audio signal includes: a left channel audio signal and a right channel audio signal; and obtaining a delayed signal corresponding to the first audio signal includes:

[0012] Delay processing is performed on the left channel audio signal and the right channel audio signal based on the first delay parameter to obtain a left channel first reflection signal and a right channel first reflection signal;

[0013] Delay processing is performed on the left channel audio signal and the right channel audio signal based on the second delay parameter to obtain a left channel second reflection signal and a right channel second reflection signal;

[0014] Linearly superimposing the left channel audio signal, the left channel first reflected signal and the left channel second reflected signal to obtain the left channel delayed signal;

[0015] The right channel audio signal, the right channel first reflected signal and the right channel second reflected signal are linearly superimposed to obtain the right channel delayed signal.

[0016] As an optional implementation of the embodiment of the present application, the delayed feedback signal corresponding to the first sampling point includes: delayed feedback signals corresponding to 2N processing channels, where N is a positive integer; and generating a reverberation signal corresponding to the first audio signal according to the delayed signal corresponding to the first audio signal and the delayed feedback signal corresponding to the first audio signal includes:

[0017] Generate input signals corresponding to each processing channel according to the left channel delayed signal and the right channel delayed signal;

[0018] For each processing channel, an input signal corresponding to the processing channel and a delayed feedback signal corresponding to the processing channel are processed by a feedback comb filter to obtain a comb filter signal corresponding to the processing channel;

[0019] The comb filter signals corresponding to the respective processing channels are input into the feedback delay network to obtain the reverberation signals corresponding to the respective processing channels output by the feedback delay network.

[0020] As an optional implementation of the embodiment of the present application, the step of generating input signals corresponding to each processing channel according to the left channel delayed signal and the right channel delayed signal includes:

[0021] Generate input signals corresponding to N processing channels according to the left channel delayed signal and the inverted signal of the left channel delayed signal;

[0022] Input signals corresponding to N processing channels are generated according to the right channel delayed signal and the inverted signal of the right channel delayed signal.

[0023] As an optional implementation of the embodiment of the present application, before processing the input signal corresponding to the processing channel and the delayed feedback signal corresponding to the processing channel through the feedback comb filter for each processing channel, the method further includes:

[0024] For each processing channel, an all-pass filtering network is used to perform an all-pass filtering process on the input signal corresponding to the processing channel;

[0025] The all-pass filtering network is an all-pass filter, or the all-pass filtering network is composed of a plurality of all-pass filters connected in series.

[0026] As an optional implementation of the embodiment of the present application, after obtaining the reverberation signal corresponding to each processing channel output by the feedback delay network, the method further includes:

[0027] Determine a first frequency band, a second frequency band, and a third frequency band of the reverberation signal corresponding to each processing channel based on the first frequency and the second frequency;

[0028] Performing attenuation processing on a first frequency band of a reverberation signal corresponding to each processing channel based on a first attenuation coefficient;

[0029] Performing attenuation processing on the second frequency band of the reverberation signal corresponding to each processing channel based on the second attenuation coefficient;

[0030] Performing attenuation processing on the third frequency band of the reverberation signal corresponding to each processing channel based on the third attenuation coefficient;

[0031] The center frequency of the first frequency band is smaller than the center frequency of the second frequency band, the center frequency of the second frequency band is smaller than the center frequency of the third frequency band, the first attenuation coefficient is smaller than the second attenuation coefficient, and the second attenuation coefficient is smaller than the third attenuation coefficient.

[0032] As an optional implementation of the embodiment of the present application, the method further includes:

[0033] Obtain a left channel reverberation signal and a right channel reverberation signal according to the reverberation signals corresponding to each processing channel;

[0034] The left channel reverberation signal and the right channel reverberation signal are superimposed and mixed based on a preset mixing matrix to obtain a stereo reverberation signal corresponding to the first audio signal.

[0035] As an optional implementation of the embodiment of the present application, obtaining the left channel reverberation signal and the right channel reverberation signal according to the feedback signals corresponding to each processing channel includes:

[0036] Feedback signals corresponding to two processing channels are randomly selected from feedback signals corresponding to various processing channels as the left channel reverberation signal and the right channel reverberation signal.

[0037] As an optional implementation of the embodiment of the present application, obtaining the left channel reverberation signal and the right channel reverberation signal according to the feedback signals corresponding to each processing channel includes:

[0038] Generate a first reverberation signal set and a second reverberation signal set according to the reverberation signals corresponding to the preprocessed signals; the first reverberation signal set and the second reverberation signal set both include feedback signals corresponding to at least two processing channels;

[0039] Performing weighted summation on each reverberation signal in the first reverberation signal set to obtain the left channel reverberation signal;

[0040] A weighted sum is performed on each reverberation signal in the second reverberation signal set to obtain the right channel reverberation signal.

[0041] As an optional implementation of the embodiment of the present application, after obtaining the stereo reverberation signal corresponding to the first audio signal, the method further includes:

[0042] linearly mixing the first audio signal and a stereo reverberation signal corresponding to the first audio signal based on a preset ratio to obtain an output audio signal corresponding to the first audio signal;

[0043] Output an output audio signal corresponding to the first audio signal.

[0044] In a second aspect, an embodiment of the present application provides an audio signal processing device, comprising:

[0045] An input module, configured to obtain a first audio signal, where the first audio signal is an audio signal corresponding to a first sampling point;

[0046] A delay module, used to obtain a delay signal corresponding to the first audio signal;

[0047] a feedback module, configured to delay a reverberation signal corresponding to a second audio signal to obtain a delayed feedback signal corresponding to the first sampling point; the second audio signal is an audio signal corresponding to a second sampling point, and the second sampling point is a sampling point before the first sampling point;

[0048] The reverberation module is configured to generate a reverberation signal corresponding to the first audio signal according to the delayed signal corresponding to the first audio signal and the delayed feedback signal corresponding to the first audio signal.

[0049] As an optional implementation of the embodiment of the present application, the first audio signal includes: a left channel audio signal and a right channel audio signal; the delay module is specifically used to delay the left channel audio signal and the right channel audio signal based on a first delay parameter to obtain a left channel first reflection signal and a right channel first reflection signal; delay the left channel audio signal and the right channel audio signal based on a second delay parameter to obtain a left channel second reflection signal and a right channel second reflection signal; linearly superimpose the left channel audio signal, the left channel first reflection signal and the left channel second reflection signal to obtain the left channel delayed signal; linearly superimpose the right channel audio signal, the right channel first reflection signal and the right channel second reflection signal to obtain the right channel delayed signal.

[0050] As an optional implementation of the embodiment of the present application, the delayed feedback signal corresponding to the first sampling point includes: delayed feedback signals corresponding to 2N processing channels, N is a positive integer; the reverberation module is specifically used to generate input signals corresponding to each processing channel according to the left channel delayed signal and the right channel delayed signal; for each processing channel, the input signal corresponding to the processing channel and the delayed feedback signal corresponding to the processing channel are processed by a feedback comb filter to obtain a comb filtered signal corresponding to the processing channel; the comb filtered signal corresponding to each processing channel is input into a feedback delay network to obtain a reverberation signal corresponding to each processing channel output by the feedback delay network.

[0051] As an optional implementation of the embodiment of the present application, the reverberation module is specifically used to generate input signals corresponding to N processing channels according to the left channel delayed signal and the inverted signal of the left channel delayed signal; and generate input signals corresponding to N processing channels according to the right channel delayed signal and the inverted signal of the right channel delayed signal.

[0052] As an optional implementation of the embodiment of the present application, the reverberation module is further used to, for each processing channel, perform all-pass filtering on the input signal corresponding to the processing channel through an all-pass filtering network before processing the input signal corresponding to the processing channel and the delayed feedback signal corresponding to the processing channel through a feedback comb filter;

[0053] The all-pass filtering network is an all-pass filter, or the all-pass filtering network is composed of a plurality of all-pass filters connected in series.

[0054] As an optional implementation of the embodiment of the present application, the reverberation module is further used to determine, based on the first frequency and the second frequency, a first frequency band, a second frequency band and a third frequency band of the reverberation signal corresponding to each processing channel after obtaining the reverberation signal corresponding to each processing channel output by the feedback delay network; perform attenuation processing on the first frequency band of the reverberation signal corresponding to each processing channel based on the first attenuation coefficient; perform attenuation processing on the second frequency band of the reverberation signal corresponding to each processing channel based on the second attenuation coefficient; perform attenuation processing on the third frequency band of the reverberation signal corresponding to each processing channel based on the third attenuation coefficient;

[0055] The center frequency of the first frequency band is smaller than the center frequency of the second frequency band, the center frequency of the second frequency band is smaller than the center frequency of the third frequency band, the first attenuation coefficient is smaller than the second attenuation coefficient, and the second attenuation coefficient is smaller than the third attenuation coefficient.

[0056] As an optional implementation of the embodiment of the present application, the reverberation module is also used to obtain a left channel reverberation signal and a right channel reverberation signal according to the reverberation signals corresponding to each processing channel; and to perform superposition and mixing processing on the left channel reverberation signal and the right channel reverberation signal based on a preset mixing matrix to obtain a stereo reverberation signal corresponding to the first audio signal.

[0057] As an optional implementation of the embodiment of the present application, the reverberation module is specifically used to randomly select feedback signals corresponding to two processing channels from feedback signals corresponding to various processing channels as the left channel reverberation signal and the right channel reverberation signal.

[0058] As an optional implementation of the embodiment of the present application, the reverberation module is specifically used to generate a first reverberation signal set and a second reverberation signal set according to the reverberation signals corresponding to the preprocessing signals; the first reverberation signal set and the second reverberation signal set both include feedback signals corresponding to at least two processing channels; weighted summation is performed on the reverberation signals in the first reverberation signal set to obtain the left channel reverberation signal; weighted summation is performed on the reverberation signals in the second reverberation signal set to obtain the right channel reverberation signal.

[0059] As an optional implementation of the embodiment of the present application, the reverberation module is further used to, after obtaining the stereo reverberation signal corresponding to the first audio signal, linearly mix the first audio signal and the stereo reverberation signal corresponding to the first audio signal based on a preset ratio to obtain an output audio signal corresponding to the first audio signal; and output the output audio signal corresponding to the first audio signal.

[0060] In a third aspect, an embodiment of the present application provides an electronic device, comprising: a memory and a processor, wherein the memory is used to store a computer program, and the processor is used to enable the electronic device to implement the audio signal processing method described in any of the above embodiments when executing the computer program.

[0061] In a fourth aspect, an embodiment of the present application provides a computer-readable storage medium, which, when the computer program is executed by a computing device, enables the computing device to implement any of the above-mentioned audio signal processing methods.

[0062] In a fifth aspect, an embodiment of the present application provides a computer program product, which, when executed on a computer, enables the computer to implement any of the above-mentioned methods for processing audio signals.

[0063] The audio signal processing method provided in the embodiment of the present application obtains the delayed signal corresponding to the first audio signal after obtaining the first audio signal corresponding to the first sampling point, delays the reverberation signal corresponding to the second audio signal, obtains the delayed feedback signal corresponding to the first sampling point, and then generates the reverberation signal corresponding to the first audio signal according to the delayed signal corresponding to the first audio signal and the delayed feedback signal corresponding to the first audio signal. Compared with the existing reverberation algorithm which only considers a small number of reflection times, since the second sampling point is the previous sampling point of the first sampling point, and when generating the reverberation signal of the first audio signal corresponding to the first sampling point, the reverberation signal of the second audio signal corresponding to the second sampling point is superimposed, and when generating the reverberation signal of the second audio signal, the reverberation signal of the audio signal corresponding to the previous sampling point is also superimposed, in the iterative processing process of the audio signals corresponding to each sampling point, when generating the reverberation signal corresponding to the current audio signal, the embodiment of the present application superimposes the reverberation signals of all audio signals before the current audio signal, so the embodiment of the present application can improve the reverberation sound effect of the sound. BRIEF DESCRIPTION OF THE DRAWINGS

[0064] 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.

[0065] In order to more clearly illustrate the technical solutions in the embodiments of the present application or the prior art, the drawings required 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.

[0066] Figure 1 A schematic diagram of a sound propagation path provided in an embodiment of the present application;

[0067] Figure 2 One of the flowcharts of the method for processing an audio signal provided in an embodiment of the present application;

[0068] Figure 3 One of the schematic diagrams of the audio signal processing device provided in the embodiment of the present application;

[0069] Figure 4 The second flowchart of the method for processing an audio signal provided in an embodiment of the present application;

[0070] Figure 5 A schematic diagram of the structure of a Schroeder reverberator provided in an embodiment of the present application;

[0071] Figure 6 A schematic diagram of the structure of a feedback delay network provided in an embodiment of the present application;

[0072] Figure 7 A second schematic diagram of the audio signal processing device provided in an embodiment of the present application;

[0073] Figure 8 A schematic diagram of a parameter input interface provided in an embodiment of the present application;

[0074] Fig. 9 A schematic diagram of the structure of an audio signal processing device provided in an embodiment of the present application;

[0075] Fig.10 A schematic diagram of the hardware structure of an electronic device provided in an embodiment of the present application. DETAILED DESCRIPTION

[0076] In order to more clearly understand the above-mentioned purposes, features and advantages of the present application, the scheme of the present application will be further described below. It should be noted that the embodiments of the present application and the features in the embodiments can be combined with each other without conflict.

[0077] In the following description, many specific details are set forth to facilitate a full understanding of the present application, but the present application may also be implemented in other ways different from those described herein. Obviously, the embodiments in the specification are only part of the embodiments of the present application, rather than all of the embodiments.

[0078] In the embodiments of the present application, words such as "exemplary" or "for example" are used to indicate examples, illustrations or descriptions. Any embodiment or design described as "exemplary" or "for example" in the embodiments of the present application should not be interpreted as being more preferred or more advantageous than other embodiments or designs. Specifically, the use of words such as "exemplary" or "for example" is intended to present related concepts in a concrete way. In addition, in the description of the embodiments of the present application, unless otherwise specified, the meaning of "multiple" refers to two or more.

[0079] Reference Figure 1 As shown, when the sound source 100 emits sound, the sound will reach the receiver 200 through multiple propagation paths. The sound source 100 may be a person, a speaker, a television, etc., and the receiver 200 may be a microphone, a person, etc. The multiple propagation paths of the sound include: the sound directly propagated from the sound source 100 to the receiver 200, the sound reaching the receiver 200 after a few reflections from the floor, wall, ceiling, etc., and the sound reaching the receiver 200 after a large number of reflections. The sound directly propagated from the sound source 100 to the receiver 200 is called direct sound, the sound reaching the receiver 200 after a small number of reflections is called early reflected sound, and the sound reaching the receiver 200 after a large number of reflections is called late reflected sound. Since the propagation paths of direct sound, early reflected sound, and late reflected sound have different lengths, the time for direct sound, early reflected sound, and late reflected sound to reach the receiver 200 is also different, and there is a certain length of delay and attenuation in the arrival of direct sound, early reflected sound, and late reflected sound, respectively, so the sound received by the receiver 200 is the reverberation sound obtained by superposition of direct sound, early reflected sound, and late reflected sound. The technical solution of the embodiment of the present application is to simulate the direct sound, early reflected sound, and late reflected sound of the original audio signal to achieve the reverberation sound effect of the sound.

[0080] The present application embodiment provides a method for processing an audio signal. Figure 2 As shown, the audio signal processing method includes the following steps S21 to S24:

[0081] S21. Acquire a first audio signal.

[0082] The first audio signal is an audio signal corresponding to a first sampling point.

[0083] The sampling point in the embodiment of the present application refers to the position where the analog audio signal is discretized and sampled in digital audio processing. Each sampling point corresponds to a specific time point and a specific audio signal value. The audio signal corresponding to the first sampling point is the audio signal value corresponding to the first sampling.

[0084] In some embodiments, the first audio signal may be a mono audio signal.

[0085] In some embodiments, the first audio signal may be a stereo audio signal including multiple channels. For example, the first audio signal may include a left channel audio signal and a right channel audio signal.

[0086] S22 obtains a delayed signal corresponding to the first audio signal.

[0087] In the embodiment of the present application, the delayed signal corresponding to the first audio signal is obtained to simulate the early reflection delay of the space. Because a delay of more than 100ms will cause the human ear to perceive the delayed sound as an echo, and less than 20ms will result in the inability to perceive the difference between the delayed sound and the direct sound, the value range of the delay parameter used when obtaining the delayed signal corresponding to the first audio signal in the embodiment of the present application can be [20ms, 100ms].

[0088] It should be noted that if more delay parameters are selected, more delay signals corresponding to the first audio signal are obtained, the reverberation density of the subsequent reverberation sound effect signal is greater, and the reverberation sound effect is better. Therefore, in some embodiments, multiple delay parameters with a spacing greater than 20sm can be selected to delay the first audio signal.

[0089] S23: Delay the reverberation signal corresponding to the second audio signal to obtain a delayed feedback signal corresponding to the first sampling point.

[0090] The second audio signal is an audio signal corresponding to a second sampling point, and the second sampling point is a sampling point before the first sampling point.

[0091] In the embodiment of the present application, the delayed feedback signal corresponding to the first sampling point is obtained to simulate the late reflection delay of the space. Since the second sampling point is the previous sampling point of the first sampling point, the reverberation signal corresponding to the second audio signal is also superimposed on the reverberation signal corresponding to the first audio signal as a late reflection signal after a certain delay and attenuation. Therefore, the embodiment of the present application first delays the reverberation signal corresponding to the second audio signal to obtain the delayed feedback signal corresponding to the first sampling point.

[0092] S24. Generate a reverberation signal corresponding to the first audio signal according to a delayed signal corresponding to the first audio signal and a delayed feedback signal corresponding to the first audio signal.

[0093] As shown above, step S22 and step S23 respectively simulate the early reflection signal (delayed signal) and the late reflection signal (delayed feedback signal) of the first audio signal. Therefore, the embodiment of the present application can obtain and generate the reverberation signal corresponding to the first audio signal through the delayed signal and the delayed feedback signal corresponding to the first audio signal.

[0094] For example, refer to Figure 3 As shown, Figure 3 FIG. 1 is a schematic diagram of the structure of an audio signal processing device for executing the audio signal processing method provided in an embodiment of the present application. Figure 3As shown, the audio signal processing device includes: a delay unit 31, a reverberation unit 32 and a feedback unit 33. The input of the audio signal processing device is the original audio signal X corresponding to each sampling point in the sampling point sequence. in The input of the audio signal processing device is the reverberation signal X corresponding to each sampling point in the sampling point sequence. out The audio signal processing flow of the audio signal processing device is as follows: first, the audio signal corresponding to the nth sampling point is input into the delay unit 31 to obtain the delayed signal of the nth sampling point, and then the audio signal corresponding to the n-1th sampling point is input into the feedback unit 33 to obtain the delayed feedback signal of the nth sampling point, and finally, the delayed signal of the nth sampling point and the delayed feedback signal of the nth sampling point are input into the reverberation unit 32 to obtain and output the reverberation signal of the nth sampling point, and when the reverberation signal of the n+1th sampling point is obtained, the reverberation signal of the nth sampling point is processed through the feedback unit 33 to obtain the delayed feedback signal of the n+1th sampling point.

[0095] The audio signal processing method provided in the embodiment of the present application obtains the delayed signal corresponding to the first audio signal after obtaining the first audio signal corresponding to the first sampling point, delays the reverberation signal corresponding to the second audio signal, obtains the delayed feedback signal corresponding to the first sampling point, and then generates the reverberation signal corresponding to the first audio signal according to the delayed signal corresponding to the first audio signal and the delayed feedback signal corresponding to the first audio signal. Compared with the existing reverberation algorithm which only considers a small number of reflection times, since the second sampling point is the previous sampling point of the first sampling point, and when generating the reverberation signal of the first audio signal corresponding to the first sampling point, the reverberation signal of the second audio signal corresponding to the second sampling point is superimposed, and when generating the reverberation signal of the second audio signal, the reverberation signal of the audio signal corresponding to the previous sampling point is also superimposed, in the iterative processing process of the audio signals corresponding to each sampling point, when generating the reverberation signal corresponding to the current audio signal, the embodiment of the present application superimposes the reverberation signals of all audio signals before the current audio signal, so the embodiment of the present application can improve the reverberation sound effect of the sound.

[0096] As an extension and refinement of the above embodiment, the present application embodiment provides another method for processing an audio signal, referring to Figure 4 As shown, the audio signal processing method includes the following steps:

[0097] S401: Acquire a first audio signal.

[0098] The first audio signal is an audio signal corresponding to a first sampling point, and the first audio signal includes: a left channel audio signal and a right channel audio signal.

[0099] Specifically, the left channel audio signal and the right channel audio signal are two components of a stereo audio signal. In stereo audio, the left channel audio signal usually transmits the audio information of the left channel, and the right channel audio signal usually transmits the audio information of the right channel. The left channel audio signal and the right channel audio signal are usually transmitted through different speakers or headphones, so that the audience can feel that the sound comes from different directions. This technology is called stereo, which can provide a more realistic and three-dimensional audio experience.

[0100] S402: Delay the left channel audio signal and the right channel audio signal based on a first delay parameter to obtain a left channel first reflection signal and a right channel first reflection signal.

[0101] The first audio signal is represented as The first delay parameter is denoted as D 1 , the first reflection signal of the left channel is expressed as The first reflection signal of the right channel is expressed as Then we have:

[0102]

[0103]

[0104] S403: Delay the left channel audio signal and the right channel audio signal based on the second delay parameter to obtain a left channel second reflection signal and a right channel second reflection signal.

[0105] The first audio signal is represented as The second delay parameter is denoted as D 2 , the second reflected signal of the left channel is expressed as The second reflection signal of the right channel is expressed as Then we have:

[0106]

[0107]

[0108] S404: Linearly superimpose the left channel audio signal, the left channel first reflected signal, and the left channel second reflected signal to obtain the left channel delayed signal.

[0109] The delayed signal of the left channel is expressed as Then we have:

[0110]

[0111] Wherein, A, B, and C are respectively the linear superposition coefficients of the left channel audio signal, the left channel first reflection signal, and the left channel second reflection signal.

[0112] In some embodiments, A, B, and C are all 1.

[0113] S405: Linearly superimpose the right channel audio signal, the right channel first reflected signal, and the right channel second reflected signal to obtain the right channel delayed signal.

[0114] The delayed signal of the left channel is expressed as Then we have:

[0115]

[0116] Wherein, D, E, and F are respectively the linear superposition coefficients of the left channel audio signal, the left channel first reflection signal, and the left channel second reflection signal.

[0117] In some embodiments, D, E, and F are all 1.

[0118] S406: Delay the reverberation signal corresponding to the second audio signal to obtain a delayed feedback signal corresponding to the first sampling point.

[0119] The second audio signal is an audio signal corresponding to a second sampling point, the second sampling point is a sampling point before the first sampling point, and the delayed feedback signal corresponding to the first sampling point includes: delayed feedback signals corresponding to 2N processing channels, where N is a positive integer.

[0120] In some embodiments, N=4. That is, the delayed feedback signal corresponding to the first sampling point includes delayed feedback signals corresponding to 8 processing channels.

[0121] In some embodiments, delaying the reverberation signal corresponding to the second audio signal includes: delaying the reverberation signal of the processing channel based on delay parameters corresponding to each processing channel to obtain a delayed feedback signal corresponding to the first sampling point, wherein the delay parameters corresponding to each processing channel are prime numbers to each other.

[0122] In some embodiments, the delay parameters corresponding to each processing channel are: 0.199f, 0.157f, 0.127f, 0.241f, 0.229f, 0.173f, 0.139f, and 0.251f.

[0123] S407: Generate input signals corresponding to respective processing channels according to the left channel delayed signal and the right channel delayed signal.

[0124] In some embodiments, input signals corresponding to respective processing channels are generated according to the left channel delayed signal and the right channel delayed signal, including: generating input signals corresponding to N processing channels according to the left channel delayed signal and an inverted signal of the left channel delayed signal; and generating input signals corresponding to N processing channels according to the right channel delayed signal and an inverted signal of the right channel delayed signal.

[0125] When N=4, the input signal v(n):L corresponding to N processing channels is generated according to the left channel delayed signal and the inverted signal of the left channel delayed signal:

[0126]

[0127] When N=4, the input signal v(n):R corresponding to N processing channels is generated according to the right channel delayed signal and the inverted signal of the right channel delayed signal:

[0128]

[0129] S408 . For each processing channel, perform all-pass filtering on the input signal corresponding to the processing channel through an all-pass filtering network.

[0130] The all-pass filter network is an all-pass filter (Allpass Filters), or the all-pass filter network is composed of a plurality of all-pass filters connected in series.

[0131] When N=4, a total of 8 processing channels are included. For each processing channel, performing all-pass filtering on the input signal corresponding to the processing channel through an all-pass filtering network may include performing all-pass filtering on the input signal corresponding to one processing channel respectively through 8 parallel all-pass filtering networks.

[0132] In some embodiments, the transfer function of the all-pass filter may be as follows:

[0133]

[0134] in, represents the signal after all-pass filtering, N is the filter delay calculated based on the sampling points, and g is the filter coefficient of the all-pass filter.

[0135] An all-pass filter is a filter that does not change the shape of the signal's frequency response but changes the phase response. The all-pass filter can simulate the reflection effect of sound passing through different material surfaces in space by performing all-pass filtering on the input signal, making the reverberation effect more natural. By adjusting the parameters of the all-pass filter, the hue and harmonic distribution of the reverberation can be controlled.

[0136] S409, for each processing channel, the input signal after all-pass filtering corresponding to the processing channel and the delayed feedback signal corresponding to the processing channel are processed by feedback comb filters (FBCF) to obtain a comb filter signal corresponding to the processing channel.

[0137] Feedback comb filter is a filter with fixed delay length and feedback coefficient. Its function is to simulate the reverberation effect caused by multiple reflections of sound in space. By adjusting the delay length and feedback coefficient, you can control parameters such as the reverberation room size, sound attenuation and number of reflections.

[0138] In some embodiments, the transfer function of the feedback comb filter may be as follows:

[0139]

[0140] in, represents the comb filter signal obtained by the feedback comb filter, N is the filter delay calculated based on the sampling points, and g is the filter coefficient of the feedback comb filter.

[0141] In the above embodiment, for each processing channel, an all-pass filtering process is performed on the input signal corresponding to the processing channel through an all-pass filtering network, and then for each processing channel, the input signal corresponding to the processing channel and the delayed feedback signal corresponding to the processing channel are processed through a feedback comb filter to obtain a comb filtering signal corresponding to the processing channel. The reverberator obtained by connecting the all-pass filtering network and the feedback comb filter in series is a Schroeder reverberator (Manfred Schroeder), so the above steps S408 and S409 can be implemented by 8 parallel Schroeder reverberators.

[0142] Reference Figure 5 As shown, in some embodiments, the Schroeder reverberator for performing the above steps S408 and S409 on each input signal includes: an all-pass filter network 51 composed of a first all-pass filter 511, a second all-pass filter 512 and a third all-pass filter 513, and a feedback comb filter 52 connected in series with the all-pass filter network.

[0143] S410, inputting the comb filter signals corresponding to the respective processing channels into the Feedback Delay Networks (FDN) to obtain the reverberation signals corresponding to the respective processing channels output by the Feedback Delay Networks.

[0144] The feedback delay network can be understood as a vectorized feedback network of the feedback comb filter. Figure 6As shown, in some embodiments, the network structure of the 4th order feedback delay network may include: a first linear superposition module 61, a first feedback module 62, a second linear superposition module 63, a second feedback module 64, a third linear superposition module 65, a third feedback module 66, a fourth linear superposition module 67, a fourth feedback module 68 and a feedback matrix 69. The order of the feedback delay network is the same as the number of processing channels. When the number of processing channels is 8, the feedback delay network is an 8th order feedback delay network. The network structure of the 8th order feedback delay network is similar to that of the 4th order feedback delay network, except that the path composed of the linear superposition module and the feedback module includes 8 paths, and the feedback matrix 69 is an 8th order feedback matrix.

[0145] In some embodiments, the feedback matrix of the feedback delay network is a Hadamard matrix, which is a square matrix in which the elements in each row and column are either 1 or -1, and the sum of the elements in each row and column is zero.

[0146] The above embodiment realizes the parallel connection of multiple comb filter signals through the feedback delay network to produce high-quality reverberation effect.

[0147] In addition, by adjusting the length of the feedback delay network and the feedback coefficient, parameters such as the reverberation room size, sound attenuation, and number of reflections can be controlled to achieve different types of reverberation effects.

[0148] The FDN network in the above embodiment can realize parallel connection of multiple processing channels to produce high-quality reverberation effects. In addition, the FDN network can control parameters such as the reverberation room size, sound attenuation and reflection times by adjusting the length of the processing channel and the feedback coefficient, thereby achieving different types of reverberation effects.

[0149] S411 , determining a first frequency band, a second frequency band, and a third frequency band of a reverberation signal corresponding to each processing channel based on the first frequency and the second frequency.

[0150] In some embodiments, the first frequency is a low-intermediate frequency crossover frequency, and the second frequency is a medium-high frequency crossover frequency. For example, if the frequency range of the reverberation signal corresponding to the processing channel is [0, F], the first frequency is F1, and the second frequency is F2, then the first frequency band, the second frequency band, and the third frequency band of the reverberation signal are [0, F1], (F1, F2], (F2, F], respectively.

[0151] S412: Perform attenuation processing on a first frequency band of a reverberation signal corresponding to each processing channel based on a first attenuation coefficient.

[0152] S413: Perform attenuation processing on the second frequency band of the reverberation signal corresponding to each processing channel based on the second attenuation coefficient.

[0153] S414: Perform attenuation processing on the third frequency band of the reverberation signal corresponding to each processing channel based on the third attenuation coefficient.

[0154] The center frequency of the first frequency band is smaller than the center frequency of the second frequency band, the center frequency of the second frequency band is smaller than the center frequency of the third frequency band, the first attenuation coefficient is smaller than the second attenuation coefficient, and the second attenuation coefficient is smaller than the third attenuation coefficient.

[0155] The above steps S411 to S414 are for simulating the attenuation characteristics of different frequencies of sound in real reverberation. In view of the fact that the higher the frequency of the sound, the greater the attenuation, and the lower the frequency of the sound, the smaller the attenuation, the embodiment of the present application first divides the reverberation signal corresponding to each processing channel into a low frequency band, a mid-frequency band and a high frequency band based on the first frequency and the second frequency, and performs attenuation processing on the sound signals in the low frequency band, the mid-frequency band and the high frequency band based on different attenuation coefficients, so that the reverberation effect is closer to the sound attenuation in the real environment, and affects the auditory perception, making the reverberation effect more natural and realistic.

[0156] In some embodiments, the device for performing the above steps S412 to S414 may be 8 damping filters connected in parallel, each damping filter being a three-bandwidth filter implemented by cascading a first-order low shelf filter and a low pass filter.

[0157] In some embodiments, the transfer function of the low shelf filter may be as follows:

[0158]

[0159]

[0160] Among them, g 0 is the DC band gain, g m is the desired band gain at the intermediate frequency.

[0161] In some embodiments, the transfer function of the low pass filter may be as follows:

[0162]

[0163] Among them, P h For the extreme point.

[0164] The transfer function of the damped filter can be expressed as:

[0165] H d (z) = H l (z)H h (z)

[0166] S415 . Obtain a left-channel reverberation signal and a right-channel reverberation signal according to the reverberation signals corresponding to the respective processing channels.

[0167] In some embodiments, obtaining a left channel reverberation signal and a right channel reverberation signal according to reverberation signals corresponding to each processing channel includes: randomly selecting feedback signals corresponding to two processing channels from feedback signals corresponding to each processing channel as the left channel reverberation signal and the right channel reverberation signal.

[0168] For example, the reverberation signal (output signal of the damping filter) corresponding to each processing channel is expressed as: 0 (n), w 1 (n),w 2 (n),…,w 7 (n)] T , two of them can be randomly selected as the left channel reverberation signal and the right channel reverberation signal. 2 (n) and w 3 (n) as the left channel reverberation signal and the right channel reverberation signal

[0169] In some embodiments, obtaining a left channel reverberation signal and a right channel reverberation signal according to reverberation signals corresponding to each processing channel includes: generating a first reverberation signal set and a second reverberation signal set according to the reverberation signals corresponding to each preprocessing signal; the first reverberation signal set and the second reverberation signal set both include feedback signals corresponding to at least two processing channels; performing weighted summation on each reverberation signal in the first reverberation signal set to obtain the left channel reverberation signal; performing weighted summation on each reverberation signal in the second reverberation signal set to obtain the right channel reverberation signal.

[0170] For example, the reverberation signal (output signal of the damping filter) corresponding to each processing channel is expressed as: 0 (n), w 1 (n),w 2 (n),...,w 7 (n)] T , then the first reverberation signal set can be generated: {w 0 (n), w 1 (n), w 2 (n), w 3 (n)} and the second reverberation signal set: {w 4 (n), w 5 (n), w 6 (n), w 7(n)}, then performing weighted summation on each reverberation signal in the first reverberation signal set to obtain the left channel reverberation signal, and performing weighted summation on each reverberation signal in the second reverberation signal set to obtain the right channel reverberation signal.

[0171] S416: Perform superposition and mixing processing on the left channel reverberation signal and the right channel reverberation signal based on a preset mixing matrix to obtain a stereo reverberation signal corresponding to the first audio signal.

[0172] In some embodiments, the preset mixing matrix may be a 2nd order Hadamard matrix. When the preset mixing matrix is ​​a 2nd order Hadamard matrix, the left channel reverberation signal is w 2 (n), the right channel reverberation signal is w 3 (n), then the second-order Hadamard matrix is ​​expressed as H 2 , the stereo reverberation signal corresponding to the first audio signal is y(n), then:

[0173]

[0174] S417: Linearly mix the first audio signal and a stereo reverberation signal corresponding to the first audio signal based on a preset ratio to obtain an output audio signal corresponding to the first audio signal.

[0175] Reference Figure 7 As shown, it is used to perform the above Figure 4 The audio signal processing device of the audio signal processing method shown includes: a pre-delay module 71, a distribution module 72, a reverberation module 73, a feedback delay network 74, a damping filter network 75, a feedback module 76 and a fusion module 77. The pre-delay module 71 is used to receive the left channel audio signal L and the right channel audio signal R of the first audio signal, and obtain the left channel delay signal Ld and the right channel delay signal Rd for the left channel audio signal L and the right channel audio signal R based on the first delay parameter and the second delay parameter. The distribution module 72 is used to process the input signals v(n):L and v(n):R corresponding to each processing channel according to the left channel delay signal Ld and the right channel delay signal Rd. The reverberation module 73 is used to obtain the comb filter signal [c] corresponding to each processing channel according to the input signals v(n):L and v(n):R corresponding to each processing channel and the delayed feedback signals corresponding to each processing channel. 0 (n),c 1 (n),c 2 (n),…,c 7 (n)] T The feedback delay network 74 is used to generate a comb filter signal corresponding to each processing channel [c 0 (n),c 1 (n),c2 (n),...,c 7 (n)] T Get the feedback delay signal corresponding to each processing channel [F 0 (n),F 1 (n),F 2 (n),...,F 7 (n)] T The damping filter network 75 is used to filter the reverberation signal corresponding to each processing channel to obtain the reverberation signal corresponding to each processing channel [w 0 (n),w 1 (n),w 2 (n),…,w 7 (n)] T The fusion module 76 is used to obtain a left channel reverberation signal and a right channel reverberation signal according to the reverberation signals corresponding to each processing channel; and to perform superposition and mixing processing on the left channel reverberation signal and the right channel reverberation signal based on a preset mixing matrix to obtain a stereo reverberation signal corresponding to the first audio signal. The feedback module 77 is used to feedback the reverberation signals corresponding to each processing channel. 0 (n),w 1 (n),w 2 (n),…,w 7 (n)] T Delay processing is performed to obtain the delayed feedback signal corresponding to the next sampling point [d 0 (n),d 1 (n),d 2 (n),…,d 7 (n)] T .

[0176] Furthermore, Figure 4 The audio signal processing method shown in the figure needs to obtain the first delay parameter, the second delay parameter, the first frequency, the second frequency, the low frequency damping filter frequency, the intermediate frequency damping filter frequency, the high frequency damping filter frequency, the dry and wet sound mixing ratio and other parameters during the audio signal processing. Figure 8As shown, the parameter input interface for receiving parameters such as the first delay parameter, the second delay parameter, the first frequency, the second frequency, the first attenuation coefficient, the second attenuation coefficient, the third attenuation coefficient, and the dry-wet sound mixing ratio includes: a delay setting area 81, wherein the delay setting area 81 includes a first control 811 for inputting the first delay parameter and a second control 812 for inputting the second delay parameter. A damping filter parameter setting area 82, wherein the damping filter parameter setting area 82 includes: a third control 821 for inputting the first frequency, a fourth control 822 for inputting the second frequency, a fifth control 823 for inputting the first attenuation coefficient, a sixth control 824 for inputting the second attenuation coefficient, and a seventh control 825 for inputting the third attenuation coefficient. A mixing ratio setting area 83, wherein the mixing ratio setting area 83 includes an eighth control 831 for setting the dry-wet sound mixing ratio. A low frequency gain setting area 84, wherein the low frequency gain setting area 84 includes: a ninth control 841 for setting the low frequency and a tenth control 842 for setting the low frequency gain. The high frequency gain setting area 85 includes an eleventh control 851 for setting the high frequency and a twelfth control 852 for setting the low frequency gain.

[0177] Based on the same inventive concept, as an implementation of the above method, an embodiment of the present application also provides an audio signal processing device, which corresponds to the above method embodiment. For ease of reading, this embodiment will no longer repeat the details of the above method embodiment one by one, but it should be clear that the audio signal processing device in this embodiment can correspond to all the contents in the above method embodiment.

[0178] The present application provides an audio signal processing device. Fig. 9 is a structural diagram of the audio signal processing device, such as Fig. 9 As shown, the audio signal processing device 900 includes:

[0179] An input module 91 is used to obtain a first audio signal, where the first audio signal is an audio signal corresponding to a first sampling point;

[0180] A delay module 92, configured to obtain a delay signal corresponding to the first audio signal;

[0181] A feedback module 93 is used to delay the reverberation signal corresponding to the second audio signal to obtain a delayed feedback signal corresponding to the first sampling point; the second audio signal is an audio signal corresponding to the second sampling point, and the second sampling point is a sampling point before the first sampling point;

[0182] The reverberation module 94 is configured to generate a reverberation signal corresponding to the first audio signal according to the delayed signal corresponding to the first audio signal and the delayed feedback signal corresponding to the first audio signal.

[0183] As an optional implementation of the embodiment of the present application, the first audio signal includes: a left channel audio signal and a right channel audio signal; the delay module 92 is specifically used to delay the left channel audio signal and the right channel audio signal based on a first delay parameter to obtain a left channel first reflection signal and a right channel first reflection signal; delay the left channel audio signal and the right channel audio signal based on a second delay parameter to obtain a left channel second reflection signal and a right channel second reflection signal; linearly superimpose the left channel audio signal, the left channel first reflection signal and the left channel second reflection signal to obtain the left channel delayed signal; linearly superimpose the right channel audio signal, the right channel first reflection signal and the right channel second reflection signal to obtain the right channel delayed signal.

[0184] As an optional implementation of the embodiment of the present application, the delayed feedback signal corresponding to the first sampling point includes: delayed feedback signals corresponding to 2N processing channels, N is a positive integer; the reverberation module 94 is specifically used to generate input signals corresponding to each processing channel according to the left channel delay signal and the right channel delay signal; for each processing channel, the input signal corresponding to the processing channel and the delayed feedback signal corresponding to the processing channel are processed by a feedback comb filter to obtain a comb filtered signal corresponding to the processing channel; the comb filtered signal corresponding to each processing channel is input into a feedback delay network to obtain a reverberation signal corresponding to each processing channel output by the feedback delay network.

[0185] As an optional implementation of the embodiment of the present application, the reverberation module 94 is specifically used to generate input signals corresponding to N processing channels according to the left channel delayed signal and the inverted signal of the left channel delayed signal; and to generate input signals corresponding to N processing channels according to the right channel delayed signal and the inverted signal of the right channel delayed signal.

[0186] As an optional implementation of the embodiment of the present application, the reverberation module 94 is further configured to, for each processing channel, perform all-pass filtering on the input signal corresponding to the processing channel through an all-pass filtering network before processing the input signal corresponding to the processing channel and the delayed feedback signal corresponding to the processing channel through a feedback comb filter;

[0187] The all-pass filtering network is an all-pass filter, or the all-pass filtering network is composed of a plurality of all-pass filters connected in series.

[0188] As an optional implementation of the embodiment of the present application, the reverberation module 94 is further used to determine, based on the first frequency and the second frequency, the first frequency band, the second frequency band and the third frequency band of the reverberation signal corresponding to each processing channel after obtaining the reverberation signal corresponding to each processing channel output by the feedback delay network; perform attenuation processing on the first frequency band of the reverberation signal corresponding to each processing channel based on the first attenuation coefficient; perform attenuation processing on the second frequency band of the reverberation signal corresponding to each processing channel based on the second attenuation coefficient; perform attenuation processing on the third frequency band of the reverberation signal corresponding to each processing channel based on the third attenuation coefficient;

[0189] The center frequency of the first frequency band is smaller than the center frequency of the second frequency band, the center frequency of the second frequency band is smaller than the center frequency of the third frequency band, the first attenuation coefficient is smaller than the second attenuation coefficient, and the second attenuation coefficient is smaller than the third attenuation coefficient.

[0190] As an optional implementation of the embodiment of the present application, the reverberation module 94 is also used to obtain a left channel reverberation signal and a right channel reverberation signal according to the reverberation signals corresponding to each processing channel; and to perform superposition and mixing processing on the left channel reverberation signal and the right channel reverberation signal based on a preset mixing matrix to obtain a stereo reverberation signal corresponding to the first audio signal.

[0191] As an optional implementation of the embodiment of the present application, the reverberation module 94 is specifically used to randomly select feedback signals corresponding to two processing channels from feedback signals corresponding to various processing channels as the left channel reverberation signal and the right channel reverberation signal.

[0192] As an optional implementation of the embodiment of the present application, the reverberation module 94 is specifically used to generate a first reverberation signal set and a second reverberation signal set according to the reverberation signals corresponding to the preprocessing signals; the first reverberation signal set and the second reverberation signal set both include feedback signals corresponding to at least two processing channels; weighted summation is performed on the reverberation signals in the first reverberation signal set to obtain the left channel reverberation signal; weighted summation is performed on the reverberation signals in the second reverberation signal set to obtain the right channel reverberation signal.

[0193] As an optional implementation of the embodiment of the present application, the reverberation module 94 is further used to linearly mix the first audio signal and the stereo reverberation signal corresponding to the first audio signal based on a preset ratio after obtaining the stereo reverberation signal corresponding to the first audio signal to obtain an output audio signal corresponding to the first audio signal; and output the output audio signal corresponding to the first audio signal.

[0194] The audio signal processing device provided in the embodiment of the present application can execute the audio signal processing method provided in any of the above embodiments, and its implementation principle and technical effect are similar, which will not be repeated here.

[0195] Based on the same inventive concept, an embodiment of the present application also provides an electronic device. Fig.10 A schematic diagram of the structure of an electronic device provided in an embodiment of the present application, such as Fig.10 As shown, the electronic device provided in this embodiment includes: a memory 101 and a processor 102, wherein the memory 101 is used to store a computer program, and the processor 102 is used to execute the audio signal processing method provided in the above embodiment when executing the computer program.

[0196] Based on the same inventive concept, an embodiment of the present application further provides a computer-readable storage medium, on which a computer program is stored. When the computer program is executed by a processor, the computing device implements the audio signal processing method provided in the above embodiment.

[0197] Based on the same inventive concept, an embodiment of the present application further provides a computer program product. When the computer program product is run on a computer, the computing device implements the audio signal processing method provided in the above embodiment.

[0198] Those skilled in the art will appreciate that the embodiments of the present application may be provided as methods, systems, or computer program products. Therefore, the present application may adopt the form of a complete hardware embodiment, a complete software embodiment, or an embodiment combining software and hardware. Moreover, the present application may adopt the form of a computer program product implemented on one or more computer-usable storage media that include computer-usable program code.

[0199] The processor may be a central processing unit (CPU), other general-purpose processors, digital signal processors (DSP), application-specific integrated circuits (ASIC), field-programmable gate arrays (FPGA) or other programmable logic devices, discrete gate or transistor logic devices, discrete hardware components, etc. A general-purpose processor may be a microprocessor or any conventional processor, etc.

[0200] The memory may include non-permanent memory in a computer-readable medium, random access memory (RAM) and / or non-volatile memory in the form of read-only memory (ROM) or flash RAM. The memory is an example of a computer-readable medium.

[0201] Computer readable media include permanent and non-permanent, removable and non-removable storage media. Storage media can be implemented by any method or technology to store information, and the information can be computer-readable instructions, data structures, program modules or other data. Examples of computer storage media include, but are not limited to, phase change memory (PRAM), static random access memory (SRAM), dynamic random access memory (DRAM), other types of random access memory (RAM), read-only memory (ROM), electrically erasable programmable read-only memory (EEPROM), flash memory or other memory technology, compact disk read-only memory (CD-ROM), digital versatile disk (DVD) or other optical storage, magnetic cassettes, magnetic disk storage or other magnetic storage devices or any other non-transmission media that can be used to store information that can be accessed by a computing device. As defined herein, computer readable media does not include temporary computer readable media (transitory media), such as modulated data signals and carrier waves.

[0202] Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present application, rather than to limit it. Although the present application has been described in detail with reference to the aforementioned embodiments, those skilled in the art should understand that they can still modify the technical solutions described in the aforementioned embodiments, or make equivalent replacements for some or all of the technical features therein, and these modifications or replacements do not cause the essence of the corresponding technical solutions to deviate from the scope of the technical solutions of the embodiments of the present application.

Claims

1. A method for processing an audio signal, It is characterized in that include: Acquire a first audio signal, where the first audio signal is an audio signal corresponding to a first sampling point; Acquire a delayed signal corresponding to the first audio signal; Delaying the reverberation signal corresponding to the second audio signal to obtain a delayed feedback signal corresponding to the first sampling point; the second audio signal is an audio signal corresponding to the second sampling point, and the second sampling point is a sampling point before the first sampling point; A reverberation signal corresponding to the first audio signal is generated according to a delayed signal corresponding to the first audio signal and a delayed feedback signal corresponding to the first audio signal.

2. The method according to claim 1, It is characterized in that The first audio signal includes: a left channel audio signal and a right channel audio signal; and the step of obtaining a delayed signal corresponding to the first audio signal includes: Delay processing is performed on the left channel audio signal and the right channel audio signal based on the first delay parameter to obtain a left channel first reflection signal and a right channel first reflection signal; Delay processing is performed on the left channel audio signal and the right channel audio signal based on the second delay parameter to obtain a left channel second reflection signal and a right channel second reflection signal; Linearly superimposing the left channel audio signal, the left channel first reflected signal and the left channel second reflected signal to obtain the left channel delayed signal; The right channel audio signal, the right channel first reflected signal and the right channel second reflected signal are linearly superimposed to obtain the right channel delayed signal.

3. The method according to claim 2, It is characterized in that The delayed feedback signal corresponding to the first sampling point includes: delayed feedback signals corresponding to 2N processing channels, where N is a positive integer; and generating a reverberation signal corresponding to the first audio signal according to the delayed signal corresponding to the first audio signal and the delayed feedback signal corresponding to the first audio signal includes: Generate input signals corresponding to each processing channel according to the left channel delayed signal and the right channel delayed signal; For each processing channel, an input signal corresponding to the processing channel and a delayed feedback signal corresponding to the processing channel are processed by a feedback comb filter to obtain a comb filter signal corresponding to the processing channel; The comb filter signals corresponding to the respective processing channels are input into the feedback delay network to obtain the reverberation signals corresponding to the respective processing channels output by the feedback delay network.

4. The method according to claim 3, It is characterized in that Generating input signals corresponding to respective processing channels according to the left channel delayed signal and the right channel delayed signal comprises: Generate input signals corresponding to N processing channels according to the left channel delayed signal and the inverted signal of the left channel delayed signal; Input signals corresponding to N processing channels are generated according to the right channel delayed signal and the inverted signal of the right channel delayed signal.

5. The method according to claim 3, It is characterized in that Before processing, for each processing channel, the input signal corresponding to the processing channel and the delayed feedback signal corresponding to the processing channel by the feedback comb filter, the method further includes: For each processing channel, an all-pass filtering process is performed on the input signal corresponding to the processing channel through an all-pass filtering network; The all-pass filtering network is an all-pass filter, or the all-pass filtering network is composed of a plurality of all-pass filters connected in series.

6. The method according to claim 3, It is characterized in that After obtaining the reverberation signals corresponding to the respective processing channels output by the feedback delay network, the method further includes: Determine a first frequency band, a second frequency band, and a third frequency band of the reverberation signal corresponding to each processing channel based on the first frequency and the second frequency; Performing attenuation processing on a first frequency band of a reverberation signal corresponding to each processing channel based on a first attenuation coefficient; Performing attenuation processing on the second frequency band of the reverberation signal corresponding to each processing channel based on the second attenuation coefficient; Performing attenuation processing on the third frequency band of the reverberation signal corresponding to each processing channel based on the third attenuation coefficient; The center frequency of the first frequency band is smaller than the center frequency of the second frequency band, the center frequency of the second frequency band is smaller than the center frequency of the third frequency band, the first attenuation coefficient is smaller than the second attenuation coefficient, and the second attenuation coefficient is smaller than the third attenuation coefficient.

7. The method according to claim 3, It is characterized in that The method further comprises: Obtain a left channel reverberation signal and a right channel reverberation signal according to the reverberation signals corresponding to each processing channel; The left channel reverberation signal and the right channel reverberation signal are superimposed and mixed based on a preset mixing matrix to obtain a stereo reverberation signal corresponding to the first audio signal.

8. The method according to claim 7, It is characterized in that The step of obtaining a left channel reverberation signal and a right channel reverberation signal according to feedback signals corresponding to each processing channel includes: Feedback signals corresponding to two processing channels are randomly selected from feedback signals corresponding to various processing channels as the left channel reverberation signal and the right channel reverberation signal.

9. The method according to claim 7, It is characterized in that The step of obtaining a left channel reverberation signal and a right channel reverberation signal according to feedback signals corresponding to each processing channel includes: Generate a first reverberation signal set and a second reverberation signal set according to the reverberation signals corresponding to the preprocessed signals; the first reverberation signal set and the second reverberation signal set both include feedback signals corresponding to at least two processing channels; Performing weighted summation on each reverberation signal in the first reverberation signal set to obtain the left channel reverberation signal; A weighted sum is performed on each reverberation signal in the second reverberation signal set to obtain the right channel reverberation signal.

10. The method according to claim 7, It is characterized in that After acquiring the stereo reverberation signal corresponding to the first audio signal, the method further includes: The first audio signal and a stereo reverberation signal corresponding to the first audio signal are linearly mixed based on a preset ratio to obtain an output audio signal corresponding to the first audio signal.

11. An audio signal processing device, It is characterized in that include: An input module, configured to obtain a first audio signal, where the first audio signal is an audio signal corresponding to a first sampling point; A delay module, used to obtain a delay signal corresponding to the first audio signal; a feedback module, configured to delay a reverberation signal corresponding to a second audio signal to obtain a delayed feedback signal corresponding to the first sampling point; the second audio signal is an audio signal corresponding to a second sampling point, and the second sampling point is a sampling point before the first sampling point; The reverberation module is configured to generate a reverberation signal corresponding to the first audio signal according to the delayed signal corresponding to the first audio signal and the delayed feedback signal corresponding to the first audio signal.

12. An electronic device, It is characterized in that include: A memory and a processor, wherein the memory is used to store a computer program, and the processor is used to enable the electronic device to implement the audio signal processing method according to any one of claims 1 to 10 when executing the computer program.

13. A computer-readable storage medium, It is characterized in that The computer-readable storage medium stores a computer program, and when the computer program is executed by a computing device, the computing device implements the audio signal processing method according to any one of claims 1 to 10.

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