A time delay calculation method and device based on complex reverberation conference room environment

Abstract

The present application discloses a method and device for calculating time delay based on a complex reverberant conference room environment. The method effectively reduces the influence of interference noise on calculation accuracy by combining an impulse transfer function corresponding to a simulated conference room environment and an acquired audio signal. Secondly, by calculating an all-pass component cepstrum signal and a minimum phase cepstrum signal, the target spectrum signal can retain the all-pass part of the impulse response and a large amount of information in the audio signal, which not only ensures the correlation of received signals between different array elements, but also has strong anti-reverberation performance, and further improves the accuracy of the calculation results compared with the existing time delay estimation algorithm.

Description

Technical Field

[0001] The present application belongs to the field of speech processing technology, and in particular relates to a delay calculation method and device based on a complex reverberation conference room environment. Background Art

[0002] With the increasing intelligence of various life devices, voice interaction has attracted more and more attention. As a key technology of voice interaction, sound source localization is widely used in various fields because it is easy to use and the positioning results are not affected by electromagnetic waves and ambient light intensity. In order to make it suitable for more complex scenarios, in recent years, more and more scholars at home and abroad have been committed to improving the performance of sound source localization systems. Their research focuses on noise resistance, positioning accuracy, real-time performance, equipment miniaturization and hardware cost. Among them, the sound source localization algorithm based on arrival time difference has a simple principle and small calculation amount, and has been widely used in real-time sound source localization.

[0003] The delay estimation algorithm is the key technology of the sound source localization algorithm based on the arrival time difference. The accuracy of the time difference estimation directly affects the accuracy of the final sound source localization. Generally, the existing delay estimation algorithm has a relatively small amount of calculation and is simple in theory, but it operates on the basis of certain assumptions, such as the random noise of the two channels is unrelated to each other. However, this assumption is often not true in reality, so the universality of this theory is low in practical applications; secondly, in real life, the environment of the conference room, which contains noise and reverberation, will also have a great impact on the accuracy of the existing delay estimation algorithm. Summary of the invention

[0004] This application aims to solve the above-mentioned existing delay estimation algorithm, which has a relatively small amount of calculation and is simple in theory, but operates on the basis of certain assumptions, such as the random noise of the two channels is unrelated to each other. However, this assumption is often not true in reality, so the universality of this theory is low in practical applications; secondly, in real life, the environment of the conference room containing noise and reverberation will have a great impact on the accuracy of the existing delay estimation algorithm. Technical defects, a delay calculation method and device based on a complex reverberant conference room environment are proposed, and its technical solution is as follows:

[0005] In a first aspect, an embodiment of the present application provides a delay calculation method based on a complex reverberation conference room environment, comprising:

[0006] Based on a preset sound source position, a preset receiver position and a preset wall reflection coefficient, a pulse transmission function corresponding to the simulated conference room environment is constructed; wherein the preset wall reflection coefficient is used to simulate the conference room environment, and the preset sound source position and the preset receiver position are determined based on a spatial rectangular coordinate system corresponding to the conference room environment;

[0007] Acquire a first audio signal, and perform a convolution operation on the first audio signal and a pulse transfer function to obtain a second audio signal;

[0008] Preprocessing the second audio signal based on preset filter parameters to obtain a third audio signal, and converting the third audio signal to obtain an initial cepstrum signal;

[0009] The initial cepstrum signal is filtered based on preset filtering parameters to obtain a minimum phase cepstrum signal, and the initial cepstrum signal and the minimum phase cepstrum signal are difference calculated to calculate an all-pass component cepstrum signal;

[0010] A target spectrum signal is obtained according to the all-pass component cepstrum signal and the minimum phase cepstrum signal, and a delay estimation value is calculated based on the target spectrum signal.

[0011] In an optional solution of the first aspect, based on a preset sound source position, a preset receiver position, and a preset wall reflection coefficient, an impulse transfer function corresponding to a simulated conference room environment is constructed, including:

[0012] Based on a preset sound source position and a preset receiver position, a virtual sound source position is simulated in a spatial rectangular coordinate system corresponding to the conference room environment;

[0013] Calculating the distance between the virtual sound source and the receiver based on the virtual sound source position and the preset receiver position, and calculating the propagation time between the virtual sound source and the receiver based on the distance between the virtual sound source and the receiver and the preset sound speed parameter;

[0014] According to the distance between the virtual sound source and the receiver, the propagation time between the virtual sound source and the receiver, and the preset wall reflection coefficient, an impulse transmission function corresponding to the simulated conference room environment is constructed.

[0015] In yet another optional solution of the first aspect, preprocessing the second audio signal based on preset filter parameters to obtain a third audio signal includes:

[0016] Based on a preset passband and a preset buffer band length, performing bandpass filtering on the second audio signal;

[0017] Based on a preset pre-emphasis parameter, pre-emphasize the second audio signal after the bandpass filtering process;

[0018] Based on a preset window function, frame-by-frame windowing processing is performed on the second audio signal after the pre-emphasis processing to obtain a third audio signal.

[0019] In yet another optional solution of the first aspect, performing conversion processing on the third audio signal to obtain an initial cepstrum signal includes:

[0020] Performing fast Fourier transform processing on the third audio signal, and performing natural logarithm calculation on the third audio signal after the fast Fourier transform processing;

[0021] Perform inverse fast Fourier transform processing on the third audio signal after natural logarithm calculation to obtain an initial cepstrum signal.

[0022] In yet another optional solution of the first aspect, obtaining a target spectrum signal according to the all-pass component cepstrum signal and the minimum phase cepstrum signal includes:

[0023] Determining a pitch period range based on a preset pitch frequency range and a signal sampling frequency corresponding to the first audio signal;

[0024] The minimum phase cepstrum signal is filtered according to the pitch period range, and the all-pass component cepstrum signal and the minimum phase cepstrum signal after filtering are summed to obtain the target cepstrum signal;

[0025] Performing fast Fourier transform processing on the target cepstrum signal, and performing exponential function calculation on the target cepstrum signal after fast Fourier transform processing; wherein the base of the exponential function is a natural constant;

[0026] The target cepstrum signal after exponential function calculation is processed by inverse fast Fourier transform to obtain the target spectrum signal.

[0027] In yet another optional solution of the first aspect, the first audio signal includes an audio receiving signal corresponding to the first array element and an audio receiving signal corresponding to the second array element, and the first array element and the second array element are in the same microphone array.

[0028] In yet another optional solution of the first aspect, calculating the delay estimation value based on the target spectrum signal includes:

[0029] Performing a conjugate calculation on the first target frequency domain signal, and performing a product calculation on the first target frequency domain signal after the conjugate calculation and the second target frequency domain signal to obtain a cross-power spectrum signal; wherein the first target frequency domain signal corresponds to the audio receiving signal corresponding to the first array element, and the second target frequency domain signal corresponds to the audio receiving signal corresponding to the second array element;

[0030] Performing weighted calculation on the cross-power spectrum signal to obtain a weighted signal, and performing inverse fast Fourier transform processing on the weighted signal and the cross-power spectrum signal to obtain a cross-correlation signal;

[0031] A delay estimate is calculated based on the cross-correlation signal.

[0032] In a second aspect, an embodiment of the present application provides a delay calculation device for simulating a complex reverberation conference room environment, comprising:

[0033] A function construction module, for constructing an impulse transmission function corresponding to a simulated conference room environment based on a preset sound source position, a preset receiver position, and a preset wall reflection coefficient; wherein the preset wall reflection coefficient is used to simulate the conference room environment, and the preset sound source position and the preset receiver position are determined based on a spatial rectangular coordinate system corresponding to the conference room environment;

[0034] A first calculation module, used for acquiring a first audio signal, and performing a convolution operation on the first audio signal and a pulse transfer function to obtain a second audio signal;

[0035] A second calculation module is used to pre-process the second audio signal based on preset filter parameters to obtain a third audio signal, and convert the third audio signal to obtain an initial cepstrum signal;

[0036] A third calculation module is used to filter the initial cepstrum signal based on preset filtering parameters to obtain a minimum phase cepstrum signal, and perform difference calculation on the initial cepstrum signal and the minimum phase cepstrum signal to calculate an all-pass component cepstrum signal;

[0037] The fourth calculation module is used to obtain a target spectrum signal according to the all-pass component cepstrum signal and the minimum phase cepstrum signal, and calculate a delay estimation value based on the target spectrum signal.

[0038] In an alternative solution of the second aspect, the function building module is used to:

[0039] Based on a preset sound source position and a preset receiver position, a virtual sound source position is simulated in a spatial rectangular coordinate system corresponding to the conference room environment;

[0040] Calculating the distance between the virtual sound source and the receiver based on the virtual sound source position and the preset receiver position, and calculating the propagation time between the virtual sound source and the receiver based on the distance between the virtual sound source and the receiver and the preset sound speed parameter;

[0041] According to the distance between the virtual sound source and the receiver, the propagation time between the virtual sound source and the receiver, and the preset wall reflection coefficient, an impulse transmission function corresponding to the simulated conference room environment is constructed.

[0042] In yet another optional solution of the second aspect, the second computing module is used to:

[0043] Based on a preset passband and a preset buffer band length, performing bandpass filtering on the second audio signal;

[0044] Based on a preset pre-emphasis parameter, pre-emphasize the second audio signal after the bandpass filtering process;

[0045] Based on a preset window function, frame-by-frame windowing processing is performed on the second audio signal after the pre-emphasis processing to obtain a third audio signal.

[0046] In yet another optional solution of the second aspect, the second computing module is further configured to:

[0047] Performing fast Fourier transform processing on the third audio signal, and performing natural logarithm calculation on the third audio signal after the fast Fourier transform processing;

[0048] Perform inverse fast Fourier transform processing on the third audio signal after natural logarithm calculation to obtain an initial cepstrum signal.

[0049] In yet another optional solution of the second aspect, the fourth computing module is used for:

[0050] Determining a pitch period range based on a preset pitch frequency range and a signal sampling frequency corresponding to the first audio signal;

[0051] The minimum phase cepstrum signal is filtered according to the pitch period range, and the all-pass component cepstrum signal and the minimum phase cepstrum signal after filtering are summed to obtain the target cepstrum signal;

[0052] Performing fast Fourier transform processing on the target cepstrum signal, and performing exponential function calculation on the target cepstrum signal after fast Fourier transform processing; wherein the base of the exponential function is a natural constant;

[0053] The target cepstrum signal after exponential function calculation is processed by inverse fast Fourier transform to obtain the target spectrum signal.

[0054] In yet another optional solution of the second aspect, the first audio signal includes an audio receiving signal corresponding to the first array element and an audio receiving signal corresponding to the second array element, and the first array element and the second array element are in the same microphone array.

[0055] In yet another optional solution of the second aspect, the fourth computing module is further used to:

[0056] Performing a conjugate calculation on the first target frequency domain signal, and performing a product calculation on the first target frequency domain signal after the conjugate calculation and the second target frequency domain signal to obtain a cross-power spectrum signal; wherein the first target frequency domain signal corresponds to the audio receiving signal corresponding to the first array element, and the second target frequency domain signal corresponds to the audio receiving signal corresponding to the second array element;

[0057] Performing weighted calculation on the cross-power spectrum signal to obtain a weighted signal, and performing inverse fast Fourier transform processing on the weighted signal and the cross-power spectrum signal to obtain a cross-correlation signal;

[0058] A delay estimate is calculated based on the cross-correlation signal.

[0059] In a third aspect, an embodiment of the present application further provides a delay calculation device for simulating a complex reverberation conference room environment, including a processor and a memory;

[0060] The processor is connected to the memory;

[0061] A memory for storing executable program codes;

[0062] The processor runs the program corresponding to the executable program code by reading the executable program code stored in the memory, so as to implement the delay calculation method for simulating a complex reverberant conference room environment provided by the first aspect of the embodiment of the present application or any one of the implementation methods of the first aspect.

[0063] In a fourth aspect, an embodiment of the present application provides a computer storage medium, which stores a computer program. The computer program includes program instructions. When the program instructions are executed by a processor, the method for calculating the delay based on a complex reverberant conference room environment provided by the first aspect of the embodiment of the present application or any one of the implementation methods of the first aspect can be implemented.

[0064] In an embodiment of the present application, when performing delay calculation in a simulated conference room environment, an impulse transfer function corresponding to the simulated conference room environment can be constructed based on a preset sound source position, a preset receiver position, and a preset wall reflection coefficient; a first audio signal is obtained, and a convolution operation is performed on the first audio signal and the impulse transfer function to obtain a second audio signal; the second audio signal is preprocessed based on preset filter parameters to obtain a third audio signal, and the third audio signal is converted to obtain an initial cepstrum signal; the initial cepstrum signal is filtered based on preset filter parameters to obtain a minimum phase cepstrum signal, and the difference between the initial cepstrum signal and the minimum phase cepstrum signal is calculated to calculate an all-pass component cepstrum signal; a target spectrum signal is obtained based on the all-pass component cepstrum signal and the minimum phase cepstrum signal, and a delay estimate is calculated based on the target spectrum signal. By combining the construction of an impulse transfer function corresponding to the simulated conference room environment and the acquired audio signal, the influence of interference noise on the calculation accuracy can be effectively reduced; secondly, by calculating the all-pass component cepstrum signal and the minimum phase cepstrum signal, the target spectrum signal can retain the all-pass part of the impulse response and a large amount of information in the audio signal, which not only ensures the correlation of the received signals between different array elements, but also has strong anti-reverberation performance, further improving the accuracy of the calculation results compared with the existing delay estimation algorithm. BRIEF DESCRIPTION OF THE DRAWINGS

[0065] In order to more clearly illustrate the technical solutions in the embodiments of the present application, the drawings required for use in the embodiments will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present application. For ordinary technicians in this field, other drawings can be obtained based on these drawings without paying creative work.

[0066] Figure 1 An overall flow chart of a delay calculation method based on a complex reverberant conference room environment provided in an embodiment of the present application;

[0067] Figure 2 A schematic diagram of the effect of a complex reverberation conference room provided in an embodiment of the present application;

[0068] Figure 3 A schematic diagram of another complex reverberation conference room provided in an embodiment of the present application;

[0069] Figure 4 A schematic diagram of the structure of a delay calculation device based on a complex reverberant conference room environment provided in an embodiment of the present application;

[0070] Figure 5 A schematic diagram of the structure of another delay calculation device based on a complex reverberant conference room environment provided in an embodiment of the present application. DETAILED DESCRIPTION

[0071] The technical solutions in the embodiments of the present application will be clearly and completely described below in conjunction with the drawings in the embodiments of the present application.

[0072] In the following introduction, the terms "first" and "second" are used for descriptive purposes only and should not be understood as indicating or implying relative importance. The following introduction provides multiple embodiments of the present application, and different embodiments can be replaced or combined, so the present application can also be considered to include all possible combinations of the same and / or different embodiments recorded. Therefore, if one embodiment includes features A, B, and C, and another embodiment includes features B and D, then the present application should also be considered to include embodiments containing one or more of all other possible combinations of A, B, C, and D, although the embodiment may not be clearly recorded in the following text.

[0073] The following description provides examples and does not limit the scope, applicability or examples set forth in the claims. Changes may be made to the functions and arrangements of the elements described without departing from the scope of the present application. Various processes or components may be appropriately omitted, substituted or added to each example. For example, the described method may be performed in an order different from the order described, and various steps may be added, omitted or combined. In addition, features described in some examples may be combined in other examples.

[0074] See also Figure 1 , Figure 1 An overall flow chart of a delay calculation method based on a complex reverberant conference room environment provided by an embodiment of the present application is shown.

[0075] like Figure 1 As shown, the delay calculation method based on the complex reverberation conference room environment may at least include the following steps:

[0076] Step 102: construct an impulse transfer function corresponding to the simulated conference room environment based on a preset sound source position, a preset receiver position, and a preset wall reflection coefficient.

[0077] In the embodiment of the present application, the delay calculation method based on the complex reverberation conference room environment can be, but is not limited to, first simulating a corresponding closed rectangular space model according to the actual conference room structure, and the size parameters of the simulated closed rectangular space model are consistent with the size parameters of the actual conference room, for example, the length of the closed rectangular space model is the same as the length of the actual conference room, the width of the closed rectangular space model is the same as the width of the actual conference room, and the height of the closed rectangular space model is the same as the height of the actual conference room; or the length, width and height ratio of the closed rectangular space model is the same as the length, width and height ratio of the actual conference room, which is not limited to this. Then, an impulse transfer function corresponding to the actual sound wave propagating through multiple reflections in the actual conference room can be constructed in the simulated closed rectangular space model, and the audio signal corresponding to different array elements of the microphone array and the impulse transfer function can be combined to calculate the audio signal containing reverberation at a specified position in the simulated conference room. It can be understood that the audio signal containing reverberation here is closer to the signal received in the actual conference room, and the influence of interference noise is minimized. Then, the audio signal containing reverberation can be analyzed and processed to calculate a more accurate delay estimate.

[0078] Specifically, when performing delay calculation in a simulated conference room environment, a corresponding closed rectangular space model is first laid out according to the structural parameters of the actual conference room. The length, width and height of the closed rectangular space model can be, but are not limited to, the same as the length, width and height of the actual conference room, and the wall reflection coefficients preset for each face in the closed rectangular space model are also the same as the reflection coefficients of each wall in the actual conference room. Here, the reflection coefficient of the wall corresponds to the structural material of the wall, and different structural materials have different reflection coefficients. For example, the reflection coefficient corresponding to the gypsum material is 0.91, the reflection coefficient of cement is 0.75, the reflection coefficient of cement mortar is 0.32, and the reflection coefficient of red brick is 0.33.

[0079] Furthermore, after the corresponding closed rectangular space model is laid out, a corresponding spatial rectangular coordinate system can be established with one vertex of the closed rectangular space as the origin, but is not limited to, and a preset sound source position and a preset receiver position are expressed in coordinate form in the spatial rectangular coordinate system. The sound source position can be understood as the position where the sound is emitted by the microphone array (the microphone array includes at least two array elements), and the receiver position can be understood as the position for receiving the sound signal emitted by the microphone array, and both the sound source position and the receiver position can be expressed by (X, Y, Z).

[0080] Furthermore, after obtaining the sound source position and the receiver position corresponding to the spatial rectangular coordinate system, the pulse transmission function corresponding to the simulated conference room environment can be constructed by combining the wall reflection coefficients corresponding to each surface of the closed rectangular space model.

[0081] As an option of the embodiment of the present application, based on a preset sound source position, a preset receiver position and a preset wall reflection coefficient, an impulse transfer function corresponding to the simulated conference room environment is constructed, including:

[0082] Based on a preset sound source position and a preset receiver position, a virtual sound source position is simulated in a spatial rectangular coordinate system corresponding to the conference room environment;

[0083] Calculating the distance between the virtual sound source and the receiver based on the virtual sound source position and the preset receiver position, and calculating the propagation time between the virtual sound source and the receiver based on the distance between the virtual sound source and the receiver and the preset sound speed parameter;

[0084] According to the distance between the virtual sound source and the receiver, the propagation time between the virtual sound source and the receiver, and the preset wall reflection coefficient, an impulse transmission function corresponding to the simulated conference room environment is constructed.

[0085] See here Figure 2 The effect diagram of a complex reverberation conference room provided by the embodiment of the present application is shown. Figure 2 As shown, the effect diagram of the complex reverberation conference room is a schematic diagram of the top view effect of the simulated closed rectangular space, that is, the rectangular area in the figure can correspond to the XOY plane in the spatial rectangular coordinate system corresponding to the closed rectangular space, and a preset sound source position and a preset receiver position are displayed in the rectangular area. The sound signal emitted by the sound source position can be transmitted to the receiver position through two paths. The first path is directly reflected to the receiver position through the right side of the rectangular area, and the second position is successively through the left and right sides of the rectangular area, and then reflected to the receiver position twice. Among them, the first path corresponds to a virtual sound source position based on the plane mirror imaging principle, and the second path also corresponds to a virtual sound source position based on the plane mirror imaging principle.

[0086] Specifically, combined Figure 2 The effect diagram of the complex reverberation conference room is shown in FIG. The lower left corner of the rectangular area in the effect diagram is taken as the origin. It is assumed that the coordinates of the sound source position in the spatial rectangular coordinate system are , the length, width and height of the simulated closed rectangular space can be expressed as ,Will It is expressed as the number of times the sound wave is reflected on both sides parallel to the YOZ plane in the spatial rectangular coordinate system, and its positive and negative signs represent the positive and negative semi-axis of the virtual sound source on the X-axis respectively. The coordinate of the corresponding virtual sound source on the X-axis can be expressed as:

[0087]

[0088]

[0089] Similarly, find The coordinates of the corresponding virtual sound source on the Y axis and on the Z axis can be expressed as:

[0090]

[0091]

[0092] In the above formula, It is expressed as the number of times the sound wave is reflected on both sides parallel to the XOZ plane in the spatial rectangular coordinate system, and its positive and negative signs represent the positive and negative semi-axis of the virtual sound source on the Y axis respectively; It is expressed as the number of times the sound wave is reflected on both sides parallel to the XOY plane in the spatial rectangular coordinate system, and its positive and negative signs represent the positive and negative half axes of the virtual sound source on the Z axis respectively.

[0093] Furthermore, after obtaining the position coordinates of the virtual sound source ( , , ), the distance between the virtual sound source and the receiver can be calculated, which can be but not limited to expressed as:

[0094]

[0095] In the above formula, the position coordinates of the receiver can be expressed as .

[0096] Furthermore, the propagation time between the virtual sound source and the receiver may be calculated according to the distance between the virtual sound source and the receiver and the preset sound speed parameter, which may be expressed as but not limited to:

[0097]

[0098] In the above formula, c may correspond to a preset sound speed parameter.

[0099] Furthermore, before constructing the pulse transfer function corresponding to the simulated conference room environment, the attenuation coefficient of the virtual sound source caused by reflection can be calculated according to the preset wall reflection coefficient. Figure 3 The effect diagram of another complex reverberation conference room provided by the embodiment of the present application is shown. Figure 3 As shown, in the simulated closed rectangular space, the reflection coefficient of the upper plane perpendicular to the Z axis in the spatial rectangular coordinate system is defined as , the reflection coefficient of the lower plane perpendicular to the Z axis in the spatial rectangular coordinate system is defined as , the reflection coefficient of the left plane perpendicular to the Y axis in the spatial rectangular coordinate system is defined as , the reflection coefficient of the right plane perpendicular to the Y axis in the spatial rectangular coordinate system is defined as , the reflection coefficient of the front plane perpendicular to the X-axis in the spatial rectangular coordinate system is defined as , the reflection coefficient of the back plane perpendicular to the X-axis in the spatial rectangular coordinate system is defined as .

[0100] Here, the attenuation coefficient of the virtual sound source due to reflection can be expressed as, but not limited to:

[0101]

[0102] It can be understood that since the attenuation of sound energy is inversely proportional to the square of the propagation distance, based on the fact that the sound pressure is inversely proportional to the distance, the amplitude coefficient corresponding to the virtual sound source can be expressed as, but not limited to:

[0103]

[0104] Furthermore, after obtaining the propagation time between the virtual sound source and the receiver and the amplitude coefficient corresponding to the virtual sound source, an impulse transfer function as shown below can be constructed but is not limited to:

[0105]

[0106] Step 104: Acquire a first audio signal, and perform a convolution operation on the first audio signal and the impulse transfer function to obtain a second audio signal.

[0107] Specifically, after obtaining the pulse transfer function, the audio signals emitted by at least two array elements in the microphone array at the above-mentioned sound source position can be received based on the above-mentioned receiver position to obtain a first audio signal corresponding to each array element, and the first audio signal and the pulse transfer function are convolved to obtain a second audio signal. Wherein, when performing the convolution operation on the first audio signal and the pulse transfer function, the first audio signal and the pulse transfer function can be, but are not limited to, input into a preset convolutional neural network, and the output result is used as the second audio signal, and the convolution operation is a common signal processing technology in the art, which is not limited here.

[0108] Step 106: pre-process the second audio signal based on preset filter parameters to obtain a third audio signal, and perform conversion processing on the third audio signal to obtain an initial cepstrum signal.

[0109] In order to ensure the quality of the audio signal, the second audio signal may be preprocessed in combination with preset filter parameters to obtain a third audio signal with higher stability and smoother sound. Here, the preset filter parameters may include, but are not limited to, a preset passband, a preset buffer band length, a maximum attenuation coefficient in the passband, a minimum attenuation coefficient in the stopband, a preset pre-emphasis parameter, and a preset window function, and are not limited thereto.

[0110] Specifically, after obtaining the second audio signal, the bandpass filter used for bandpass filtering processing can be set by using, but not limited to, an indirect method, and the preset passband is set to the voice frequency range, for example, it can be set to 300Hz-3400Hz, and then the order of the bandpass filter and the 3dB cutoff frequency are determined according to the preset buffer band length, the maximum attenuation coefficient in the passband, and the minimum attenuation coefficient in the stopband, so as to obtain a bandpass filter that meets the bandpass filtering processing requirements. It is understandable that after setting the bandpass filter that meets the bandpass filtering processing requirements, the second audio signal can be bandpass filtered in combination with the bandpass filter.

[0111] Furthermore, after the second audio signal is subjected to bandpass filtering, in order to enhance the high frequency components of the signal and make the spectrum of the signal flatter, the second audio signal subjected to bandpass filtering may be subjected to pre-emphasis processing based on a preset pre-emphasis parameter. When performing the pre-emphasis processing, the pre-emphasis digital filter (also understood as a high-pass digital filter) may be set in combination with a preset order (usually set to 1) and a preset pre-emphasis parameter. The system function of the pre-emphasis digital filter may be, but is not limited to, expressed as:

[0112]

[0113] In the above formula, The corresponding parameter is the pre-emphasis parameter, which generally takes a value between 0.9 and 1.

[0114] Based on this, the second audio signal is pre-emphasized according to the above-mentioned pre-emphasis digital filter, which can be expressed as but not limited to:

[0115]

[0116] In the above formula, can correspond to the second audio signal after pre-emphasis processing, IZT[ ] can correspond to the inverse z-transformation function, It may correspond to the second audio signal after bandpass filtering.

[0117] Since the characteristics of speech signals change slowly over time, if the statistical observation time is relatively short, it can be considered as a stationary signal. Based on this, after the second audio signal is pre-emphasized, a filter can be set based on a preset window function to divide the second audio signal after pre-emphasis into short-term stationary signals frame by frame, that is, the third audio signal. The preset window function can be, but is not limited to, a rectangular window, a Hanning window, a Hamming window, a Gaussian window, a Kaiser-Bessel window, etc.

[0118] In order to make the speech signal smoother after framing, the framing process generally causes some signal overlap between adjacent frames. Here, the frame length is set to N and the frame shift is , the second audio signal after pre-emphasis processing The length is , then the starting position of the i-th frame data can be expressed as but not limited to:

[0119]

[0120] The number of speech frames can be expressed as, but not limited to:

[0121]

[0122] Based on this, assume that the window function is expressed as , then the audio signal of the i-th frame can be expressed as but not limited to:

[0123]

[0124] Here, Can correspond to the audio signal of the i-th frame, It can be corresponded to a window function, It may correspond to the second audio signal after pre-emphasis processing.

[0125] It can be understood that the spectrum resolution of the third audio signal after the frame splitting and windowing process can be, but is not limited to, expressed as:

[0126]

[0127] In the above formula, It can correspond to the sampling period, and N can correspond to the window length.

[0128] As another option of the embodiment of the present application, converting the third audio signal to obtain an initial cepstrum signal includes:

[0129] Performing fast Fourier transform processing on the third audio signal, and performing natural logarithm calculation on the third audio signal after the fast Fourier transform processing;

[0130] Perform inverse fast Fourier transform processing on the third audio signal after natural logarithm calculation to obtain an initial cepstrum signal.

[0131] Specifically, after the third audio signal is obtained through preprocessing, the third audio signal may be transformed to obtain a corresponding initial cepstrum signal. Here, the third audio signal includes an audio signal corresponding to the first array element of the microphone array and an audio signal corresponding to the second array element of the microphone array as an example. Assume that the audio signal corresponding to the first array element of the microphone array can be expressed as , the audio signal corresponding to the second element of the microphone array can be expressed as ,but as well as After fast Fourier transform processing, the corresponding signals are obtained as well as .

[0132] Then, we can respectively as well as Perform natural logarithm calculation, the calculation expression of the natural logarithm can be but is not limited to , and perform inverse fast Fourier transform on the signal after natural logarithm calculation to obtain the initial cepstrum signal as well as .

[0133] Step 108 : filtering the initial cepstrum signal based on preset filtering parameters to obtain a minimum phase cepstrum signal, and performing difference calculation on the initial cepstrum signal and the minimum phase cepstrum signal to calculate an all-pass component cepstrum signal.

[0134] After obtaining the initial cepstrum signal, the initial cepstrum signal can be filtered based on a preset filtering parameter to obtain a cepstrum signal of a minimum phase component, wherein the preset filtering parameter can be, but is not limited to, represented as The filtering processing method can be but is not limited to multiplying the preset filtering parameters with the initial cepstrum signal. It can be understood that the filtering processing method is a common signal filtering technology in the art and will not be described in detail here.

[0135] Here, the initial cepstrum signal mentioned above is as well as For example, the cepstrum signal of the minimum phase component obtained after filtering can be expressed as but not limited to: as well as .

[0136] Furthermore, after obtaining the cepstrum signal of the minimum phase component, the initial cepstrum signal and the minimum phase cepstrum signal may be subjected to difference calculation to calculate the cepstrum signal of the full-pass component. Here, the cepstrum signal of the minimum phase component mentioned above may be, but is not limited to, respectively expressed as as well as , the initial cepstrum signals are as well as As an example, the cepstrum signals of the all-pass components can be, but are not limited to, respectively expressed as as well as .

[0137] Step 110: Obtain a target spectrum signal according to the all-pass component cepstrum signal and the minimum phase cepstrum signal, and calculate a delay estimation value based on the target spectrum signal.

[0138] Since the cepstrum signal of the minimum phase component includes the glottis, vocal tract part of the sound source signal and the minimum phase part of the room impulse response, which is mainly distributed near the origin in the cepstrum, while the voiced part of the sound source is distributed in the high time interval of the cepstrum, the two can be distinguished according to the position of the fundamental period on the cepstrum.

[0139] Specifically, after obtaining the cepstrum signal of the all-pass component and the cepstrum signal of the minimum phase component, the pitch period range can be determined based on the preset pitch frequency range and the signal sampling frequency corresponding to the first audio signal. Here, the preset pitch frequency range is expressed as , the signal sampling frequency corresponding to the first audio signal is expressed as For example, the pitch period range can be, but is not limited to, expressed as follows:

[0140]

[0141] Further, after obtaining the pitch period range, different value ranges can be set according to the pitch period range, and the cepstrum signal of the minimum phase component can be filtered in combination with different value ranges to filter out the cepstrum signal of the minimum phase component with less interference noise, which can be expressed as but not limited to:

[0142]

[0143] In the above formula, The value must be larger than the pitch period range, and N is the window length mentioned above.

[0144] Further, after obtaining the cepstrum signal of the minimum phase component with less interference noise, the cepstrum signal of the minimum phase component with less interference noise and the cepstrum signal of the all-pass component can be summed to obtain a target cepstrum signal, which can be expressed as but not limited to:

[0145]

[0146] In the above formula, Can correspond to the target cepstrum signal, It can be corresponded to the all-pass component cepstrum signal, It can correspond to the cepstrum signal of the minimum phase component.

[0147] Further, after obtaining the target cepstrum signal, the target cepstrum signal may be subjected to fast Fourier transform processing, and then the target cepstrum signal after fast Fourier transform processing may be subjected to exponential function calculation. The expression for exponential function calculation may be, but is not limited to, , and the calculated target cepstrum signal can be processed by inverse fast Fourier transform to convert the cepstrum signal into a spectrum signal, thereby obtaining the target spectrum signal.

[0148] Furthermore, after the target spectrum signal is obtained, the time delay estimation values ​​at the positions corresponding to the different array elements received at the receiver position may be calculated according to the target spectrum signals corresponding to the different array elements.

[0149] Here, the target spectrum signals corresponding to different array elements are respectively the first target frequency domain signal and the second target frequency domain signal. The first target frequency domain signal is expressed as , the second target frequency domain signal is expressed as , the conjugate calculation can be performed on the first target frequency domain signal to obtain , then the cross power spectrum signal can be calculated according to but not limited to the following formula:

[0150]

[0151] Furthermore, the cross power spectrum signal may be weighted according to, but not limited to, the following formula to obtain a weighted signal:

[0152]

[0153] Furthermore, the weighted signal and the cross-power spectrum signal obtained by weighted calculation may be subjected to inverse fast Fourier transform processing according to the following formula, but is not limited to the following formula, to obtain a cross-correlation signal:

[0154]

[0155] In the above formula, IFFT[ ] can correspond to the Fourier transform function.

[0156] Furthermore, an independent variable corresponding to the peak value in the cross-correlation signal may be determined to obtain an actual delay estimation value according to the independent variable, and the calculation method may be but is not limited to:

[0157]

[0158] In the above formula, arc[ ] can be corresponded to an inverse trigonometric function.

[0159] In the above-mentioned embodiment of the present application, the minimum phase part of the room impulse response can be filtered out, and the full-pass part of the room impulse response and a large amount of information in the sound source signal can be retained. Not only can the correlation of the received signals between different array elements be guaranteed, it has a strong anti-reverberation performance, and the delay error calculated by the ordinary delay estimation algorithm in the indoor environment is lower.

[0160] See also Figure 4 , Figure 4 A schematic diagram of the structure of a delay calculation device based on a complex reverberant conference room environment provided in an embodiment of the present application is shown.

[0161] like Figure 4 As shown, the delay calculation device based on a complex reverberation conference room environment may include at least a function construction module 401, a first calculation module 402, a second calculation module 403, a third calculation module 404 and a fourth calculation module 405, wherein:

[0162] Function construction module 401, used to construct an impulse transmission function corresponding to the simulated conference room environment based on a preset sound source position, a preset receiver position and a preset wall reflection coefficient; wherein the preset wall reflection coefficient is used to simulate the conference room environment, and the preset sound source position and the preset receiver position are determined based on a spatial rectangular coordinate system corresponding to the conference room environment;

[0163] A first calculation module 402 is used to obtain a first audio signal, and perform a convolution operation on the first audio signal and a pulse transfer function to obtain a second audio signal;

[0164] The second calculation module 403 is used to pre-process the second audio signal based on preset filter parameters to obtain a third audio signal, and convert the third audio signal to obtain an initial cepstrum signal;

[0165] The third calculation module 404 is used to filter the initial cepstrum signal based on preset filtering parameters to obtain a minimum phase cepstrum signal, and perform difference calculation on the initial cepstrum signal and the minimum phase cepstrum signal to calculate an all-pass component cepstrum signal;

[0166] The fourth calculation module 405 is used to obtain a target spectrum signal according to the all-pass component cepstrum signal and the minimum phase cepstrum signal, and calculate a delay estimation value based on the target spectrum signal.

[0167] In some possible embodiments, the function building block is used to:

[0168] Based on a preset sound source position and a preset receiver position, a virtual sound source position is simulated in a spatial rectangular coordinate system corresponding to the conference room environment;

[0169] Calculating the distance between the virtual sound source and the receiver based on the virtual sound source position and the preset receiver position, and calculating the propagation time between the virtual sound source and the receiver based on the distance between the virtual sound source and the receiver and the preset sound speed parameter;

[0170] According to the distance between the virtual sound source and the receiver, the propagation time between the virtual sound source and the receiver, and the preset wall reflection coefficient, an impulse transmission function corresponding to the simulated conference room environment is constructed.

[0171] In some possible embodiments, the second computing module is used to:

[0172] Based on a preset passband and a preset buffer band length, performing bandpass filtering on the second audio signal;

[0173] Based on a preset pre-emphasis parameter, pre-emphasize the second audio signal after the bandpass filtering process;

[0174] Based on a preset window function, frame-by-frame windowing processing is performed on the second audio signal after the pre-emphasis processing to obtain a third audio signal.

[0175] In some possible embodiments, the second computing module is further used for:

[0176] Performing fast Fourier transform processing on the third audio signal, and performing natural logarithm calculation on the third audio signal after the fast Fourier transform processing;

[0177] Perform inverse fast Fourier transform processing on the third audio signal after natural logarithm calculation to obtain an initial cepstrum signal.

[0178] In some possible embodiments, the fourth computing module is used to:

[0179] Determining a pitch period range based on a preset pitch frequency range and a signal sampling frequency corresponding to the first audio signal;

[0180] The minimum phase cepstrum signal is filtered according to the pitch period range, and the all-pass component cepstrum signal and the minimum phase cepstrum signal after filtering are summed to obtain the target cepstrum signal;

[0181] Performing fast Fourier transform processing on the target cepstrum signal, and performing exponential function calculation on the target cepstrum signal after fast Fourier transform processing; wherein the base of the exponential function is a natural constant;

[0182] The target cepstrum signal after exponential function calculation is processed by inverse fast Fourier transform to obtain the target spectrum signal.

[0183] In some possible embodiments, the first audio signal includes an audio receiving signal corresponding to a first array element and an audio receiving signal corresponding to a second array element, and the first array element and the second array element are in the same microphone array.

[0184] In some possible embodiments, the fourth calculation module is further used for:

[0185] Performing a conjugate calculation on the first target frequency domain signal, and performing a product calculation on the first target frequency domain signal after the conjugate calculation and the second target frequency domain signal to obtain a cross-power spectrum signal; wherein the first target frequency domain signal corresponds to the audio receiving signal corresponding to the first array element, and the second target frequency domain signal corresponds to the audio receiving signal corresponding to the second array element;

[0186] Performing weighted calculation on the cross-power spectrum signal to obtain a weighted signal, and performing inverse fast Fourier transform processing on the weighted signal and the cross-power spectrum signal to obtain a cross-correlation signal;

[0187] A delay estimate is calculated based on the cross-correlation signal.

[0188] Those skilled in the art can clearly understand that the technical solutions of the embodiments of the present application can be implemented with the help of software and / or hardware. The "unit" and "module" in this specification refer to software and / or hardware that can independently complete or cooperate with other components to complete specific functions, where the hardware can be, for example, a field programmable gate array (FPGA), an integrated circuit (IC), etc.

[0189] See also Figure 5 , Figure 5 A schematic diagram of the structure of another delay calculation device based on a complex reverberant conference room environment provided in an embodiment of the present application is shown.

[0190] like Figure 5 As shown, the delay calculation device 500 based on a complex reverberation conference room environment may include at least one processor 501 , at least one network interface 504 , a user interface 503 , a memory 505 and at least one communication bus 502 .

[0191] The communication bus 502 may be used to realize the connection and communication among the above-mentioned components.

[0192] The user interface 503 may include buttons, and the optional user interface may also include a standard wired interface or a wireless interface.

[0193] The network interface 504 may include, but is not limited to, a Bluetooth module, an NFC module, a Wi-Fi module, etc.

[0194] Among them, the processor 501 may include one or more processing cores. The processor 501 uses various interfaces and lines to connect various parts in the delay calculation device 500 based on a complex reverberation conference room environment, and executes various functions and processes data of the delay calculation device 500 based on a complex reverberation conference room environment by running or executing instructions, programs, code sets or instruction sets stored in the memory 505, and calling data stored in the memory 505. Optionally, the processor 501 can be implemented in at least one hardware form of DSP, FPGA, and PLA. The processor 501 can integrate one or a combination of CPU, GPU, modem, etc. Among them, the CPU mainly processes the operating system, user interface, and application programs; the GPU is responsible for rendering and drawing the content to be displayed on the display screen; the modem is used to process wireless communications. It can be understood that the above-mentioned modem may not be integrated into the processor 501, and it can be implemented separately through a chip.

[0195] The memory 505 may include a RAM or a ROM. Optionally, the memory 505 includes a non-transitory computer-readable medium. The memory 505 may be used to store instructions, programs, codes, code sets or instruction sets. The memory 505 may include a program storage area and a data storage area, wherein the program storage area may store instructions for implementing an operating system, instructions for at least one function (such as a touch function, a sound playback function, an image playback function, etc.), instructions for implementing the above-mentioned method embodiments, etc.; the data storage area may store data involved in the above-mentioned method embodiments, etc. The memory 505 may optionally be at least one storage device located away from the aforementioned processor 501. As Figure 5 As shown, the memory 505 as a computer storage medium may include an operating system, a network communication module, a user interface module, and a delay calculation application based on a complex reverberant conference room environment.

[0196] Specifically, the processor 501 may be used to call the delay calculation application based on the complex reverberation conference room environment stored in the memory 505, and specifically perform the following operations:

[0197] Based on a preset sound source position, a preset receiver position and a preset wall reflection coefficient, a pulse transmission function corresponding to the simulated conference room environment is constructed; wherein the preset wall reflection coefficient is used to simulate the conference room environment, and the preset sound source position and the preset receiver position are determined based on a spatial rectangular coordinate system corresponding to the conference room environment;

[0198] Acquire a first audio signal, and perform a convolution operation on the first audio signal and a pulse transfer function to obtain a second audio signal;

[0199] Preprocessing the second audio signal based on preset filter parameters to obtain a third audio signal, and converting the third audio signal to obtain an initial cepstrum signal;

[0200] The initial cepstrum signal is filtered based on preset filtering parameters to obtain a minimum phase cepstrum signal, and the initial cepstrum signal and the minimum phase cepstrum signal are difference calculated to calculate an all-pass component cepstrum signal;

[0201] A target spectrum signal is obtained according to the all-pass component cepstrum signal and the minimum phase cepstrum signal, and a delay estimation value is calculated based on the target spectrum signal.

[0202] In some possible embodiments, based on a preset sound source position, a preset receiver position, and a preset wall reflection coefficient, an impulse transfer function corresponding to a simulated conference room environment is constructed, including:

[0203] Based on a preset sound source position and a preset receiver position, a virtual sound source position is simulated in a spatial rectangular coordinate system corresponding to the conference room environment;

[0204] Calculating the distance between the virtual sound source and the receiver based on the virtual sound source position and the preset receiver position, and calculating the propagation time between the virtual sound source and the receiver based on the distance between the virtual sound source and the receiver and the preset sound speed parameter;

[0205] According to the distance between the virtual sound source and the receiver, the propagation time between the virtual sound source and the receiver, and the preset wall reflection coefficient, an impulse transmission function corresponding to the simulated conference room environment is constructed.

[0206] In some possible embodiments, preprocessing the second audio signal based on preset filter parameters to obtain the third audio signal includes:

[0207] Based on a preset passband and a preset buffer band length, performing bandpass filtering on the second audio signal;

[0208] Based on a preset pre-emphasis parameter, pre-emphasize the second audio signal after the bandpass filtering process;

[0209] Based on a preset window function, frame-by-frame windowing processing is performed on the second audio signal after the pre-emphasis processing to obtain a third audio signal.

[0210] In some possible embodiments, performing conversion processing on the third audio signal to obtain an initial cepstrum signal includes:

[0211] Performing fast Fourier transform processing on the third audio signal, and performing natural logarithm calculation on the third audio signal after the fast Fourier transform processing;

[0212] Perform inverse fast Fourier transform processing on the third audio signal after natural logarithm calculation to obtain an initial cepstrum signal.

[0213] In some possible embodiments, obtaining a target spectrum signal according to the all-pass component cepstrum signal and the minimum phase cepstrum signal includes:

[0214] Determining a pitch period range based on a preset pitch frequency range and a signal sampling frequency corresponding to the first audio signal;

[0215] The minimum phase cepstrum signal is filtered according to the pitch period range, and the all-pass component cepstrum signal and the minimum phase cepstrum signal after filtering are summed to obtain the target cepstrum signal;

[0216] Performing fast Fourier transform processing on the target cepstrum signal, and performing exponential function calculation on the target cepstrum signal after fast Fourier transform processing; wherein the base of the exponential function is a natural constant;

[0217] The target cepstrum signal after exponential function calculation is processed by inverse fast Fourier transform to obtain the target spectrum signal.

[0218] In some possible embodiments, the first audio signal includes an audio receiving signal corresponding to a first array element and an audio receiving signal corresponding to a second array element, and the first array element and the second array element are in the same microphone array.

[0219] In some possible embodiments, calculating the delay estimation value based on the target spectrum signal includes:

[0220] Performing a conjugate calculation on the first target frequency domain signal, and performing a product calculation on the first target frequency domain signal after the conjugate calculation and the second target frequency domain signal to obtain a cross-power spectrum signal; wherein the first target frequency domain signal corresponds to the audio receiving signal corresponding to the first array element, and the second target frequency domain signal corresponds to the audio receiving signal corresponding to the second array element;

[0221] Performing weighted calculation on the cross-power spectrum signal to obtain a weighted signal, and performing inverse fast Fourier transform processing on the weighted signal and the cross-power spectrum signal to obtain a cross-correlation signal;

[0222] A delay estimate is calculated based on the cross-correlation signal.

[0223] The present application also provides a computer-readable storage medium on which a computer program is stored, and when the program is executed by a processor, the steps of the above method are implemented. The computer-readable storage medium may include, but is not limited to, any type of disk, including a floppy disk, an optical disk, a DVD, a CD-ROM, a micro drive, and a magneto-optical disk, a ROM, a RAM, an EPROM, an EEPROM, a DRAM, a VRAM, a flash memory device, a magnetic card or an optical card, a nanosystem (including a molecular memory IC), or any type of medium or device suitable for storing instructions and / or data.

[0224] It should be noted that, for the aforementioned method embodiments, for the sake of simplicity, they are all expressed as a series of action combinations, but those skilled in the art should be aware that the present application is not limited by the described order of actions, because according to the present application, certain steps can be performed in other orders or simultaneously. Secondly, those skilled in the art should also be aware that the embodiments described in the specification are all preferred embodiments, and the actions and modules involved are not necessarily required by the present application.

[0225] In the above embodiments, the description of each embodiment has its own emphasis. For parts that are not described in detail in a certain embodiment, reference can be made to the relevant descriptions of other embodiments.

[0226] In the several embodiments provided in the present application, it should be understood that the disclosed devices can be implemented in other ways. For example, the device embodiments described above are only schematic, such as the division of units, which is only a logical function division. There may be other division methods in actual implementation, such as multiple units or components can be combined or integrated into another system, or some features can be ignored or not executed. Another point is that the mutual coupling or direct coupling or communication connection shown or discussed can be through some service interfaces, and the indirect coupling or communication connection of devices or units can be electrical or other forms.

[0227] The units described as separate components may or may not be physically separated, and the components shown as units may or may not be physical units, that is, they may be located in one place or distributed on multiple network units. Some or all of the units may be selected according to actual needs to achieve the purpose of the solution of this embodiment.

[0228] In addition, each functional unit in each embodiment of the present application may be integrated into one processing unit, or each unit may exist physically separately, or two or more units may be integrated into one unit. The above-mentioned integrated unit may be implemented in the form of hardware or in the form of software functional units.

[0229] If the integrated unit is implemented in the form of a software functional unit and sold or used as an independent product, it can be stored in a computer-readable memory. Based on this understanding, the technical solution of the present application, or the part that contributes to the prior art, or all or part of the technical solution can be embodied in the form of a software product. The computer software product is stored in a memory, including a number of instructions to enable a computer device (which can be a personal computer, server or network device, etc.) to execute all or part of the steps of the various embodiments of the present application. The aforementioned memory includes: U disk, read-only memory (ROM), random access memory (RAM), mobile hard disk, disk or optical disk and other media that can store program codes.

[0230] A person skilled in the art may understand that all or part of the steps in the various methods of the above embodiments may be completed by entering a program to instruct related hardware. The program may be stored in a computer-readable memory, and the memory may include: a flash drive, a read-only memory (ROM), a random access memory (RAM), a magnetic disk or an optical disk, etc.

[0231] The above are only exemplary embodiments of the present disclosure and cannot be used to limit the scope of the present disclosure. That is, any equivalent changes and modifications made according to the teachings of the present disclosure are still within the scope of the present disclosure. After considering the specification and practicing the disclosure here, those skilled in the art will easily think of the implementation scheme of the present disclosure. This application is intended to cover any modification, use or adaptation of the present disclosure, which follows the general principles of the present disclosure and includes common knowledge or customary technical means in the technical field that are not recorded in the present disclosure. The description and examples are only regarded as exemplary, and the scope and spirit of the present disclosure are defined by the claims.

Claims

1. A delay calculation method based on a complex reverberation conference room environment, characterized in that: include: Based on a preset sound source position, a preset receiver position and a preset wall reflection coefficient, an impulse transfer function corresponding to a simulated conference room environment is constructed; wherein the preset wall reflection coefficient is used to simulate the conference room environment, and the preset sound source position and the preset receiver position are determined based on a spatial rectangular coordinate system corresponding to the conference room environment; Acquire a first audio signal, and perform a convolution operation on the first audio signal and the pulse transfer function to obtain a second audio signal; wherein the first audio signal is obtained based on the preset sound source position and the preset receiver position; Preprocessing the second audio signal based on preset filter parameters to obtain a third audio signal, and converting the third audio signal to obtain an initial cepstrum signal; The initial cepstrum signal is filtered based on preset filtering parameters to obtain a minimum phase cepstrum signal, and a difference calculation is performed on the initial cepstrum signal and the minimum phase cepstrum signal to calculate an all-pass component cepstrum signal; Obtaining a target spectrum signal according to the all-pass component cepstrum signal and the minimum phase cepstrum signal, and calculating a delay estimation value based on the target spectrum signal; The step of constructing an impulse transfer function corresponding to the simulated conference room environment based on a preset sound source position, a preset receiver position, and a preset wall reflection coefficient includes: Based on a preset sound source position and a preset receiver position, a virtual sound source position is simulated in a spatial rectangular coordinate system corresponding to the conference room environment; Calculating the distance between the virtual sound source and the receiver according to the virtual sound source position and the preset receiver position, and calculating the propagation time between the virtual sound source and the receiver based on the distance between the virtual sound source and the receiver and the preset sound speed parameter; According to the distance between the virtual sound source and the receiver, the propagation time between the virtual sound source and the receiver, and a preset wall reflection coefficient, an impulse transmission function corresponding to the simulated conference room environment is constructed.

2. The method according to claim 1, characterized in that The preprocessing of the second audio signal based on the preset filter parameters to obtain the third audio signal includes: Based on a preset passband and a preset buffer band length, performing bandpass filtering on the second audio signal; Based on a preset pre-emphasis parameter, performing pre-emphasis processing on the second audio signal after the bandpass filtering processing; Based on a preset window function, frame-by-frame windowing processing is performed on the second audio signal after the pre-emphasis processing to obtain a third audio signal.

3. The method according to claim 2, characterized in that The converting process of the third audio signal to obtain an initial cepstrum signal includes: Performing fast Fourier transform processing on the third audio signal, and performing natural logarithm calculation on the third audio signal after the fast Fourier transform processing; Perform inverse fast Fourier transform processing on the third audio signal after natural logarithm calculation to obtain an initial cepstrum signal.

4. The method according to claim 1, characterized in that The step of obtaining a target spectrum signal according to the all-pass component cepstrum signal and the minimum phase cepstrum signal comprises: Determining a pitch period range based on a preset pitch frequency range and a signal sampling frequency corresponding to the first audio signal; Filtering the minimum phase cepstrum signal according to the pitch period range, and summing the all-pass component cepstrum signal and the minimum phase cepstrum signal after filtering to obtain a target cepstrum signal; Performing fast Fourier transform processing on the target cepstrum signal, and performing exponential function calculation on the target cepstrum signal after the fast Fourier transform processing; wherein the base of the exponential function is a natural constant; The target cepstrum signal after the exponential function calculation is processed by inverse fast Fourier transform to obtain a target spectrum signal.

5. The method according to claim 4, characterized in that The first audio signal includes an audio receiving signal corresponding to a first array element and an audio receiving signal corresponding to a second array element, and the first array element and the second array element are in the same microphone array.

6. The method according to claim 5, characterized in that The calculating the delay estimation value based on the target spectrum signal includes: Performing a conjugate calculation on the first target frequency domain signal, and performing a product calculation on the first target frequency domain signal and the second target frequency domain signal after the conjugate calculation to obtain a cross-power spectrum signal; wherein the first target frequency domain signal corresponds to the audio receiving signal corresponding to the first array element, and the second target frequency domain signal corresponds to the audio receiving signal corresponding to the second array element; Performing weighted calculation on the cross-power spectrum signal to obtain a weighted signal, and performing inverse fast Fourier transform processing on the weighted signal and the cross-power spectrum signal to obtain a cross-correlation signal; A delay estimate is calculated based on the cross-correlation signal.

7. A delay calculation device based on a complex reverberation conference room environment, characterized in that: include: A function construction module, used to construct an impulse transmission function corresponding to the simulated conference room environment based on a preset sound source position, a preset receiver position and a preset wall reflection coefficient; wherein the preset wall reflection coefficient is used to simulate the conference room environment, and the preset sound source position and the preset receiver position are determined based on a spatial rectangular coordinate system corresponding to the conference room environment; A first calculation module, configured to obtain a first audio signal, and perform a convolution operation on the first audio signal and the pulse transfer function to obtain a second audio signal; wherein the first audio signal is obtained based on the preset sound source position and the preset receiver position; a second calculation module, configured to preprocess the second audio signal based on preset filter parameters to obtain a third audio signal, and perform conversion processing on the third audio signal to obtain an initial cepstrum signal; A third calculation module is used to filter the initial cepstrum signal based on preset filtering parameters to obtain a minimum phase cepstrum signal, and perform difference calculation on the initial cepstrum signal and the minimum phase cepstrum signal to calculate an all-pass component cepstrum signal; a fourth calculation module, configured to obtain a target spectrum signal according to the all-pass component cepstrum signal and the minimum phase cepstrum signal, and calculate a delay estimation value based on the target spectrum signal; The step of constructing an impulse transfer function corresponding to the simulated conference room environment based on a preset sound source position, a preset receiver position, and a preset wall reflection coefficient includes: Based on a preset sound source position and a preset receiver position, a virtual sound source position is simulated in a spatial rectangular coordinate system corresponding to the conference room environment; Calculating the distance between the virtual sound source and the receiver according to the virtual sound source position and the preset receiver position, and calculating the propagation time between the virtual sound source and the receiver based on the distance between the virtual sound source and the receiver and the preset sound speed parameter; According to the distance between the virtual sound source and the receiver, the propagation time between the virtual sound source and the receiver, and a preset wall reflection coefficient, an impulse transmission function corresponding to the simulated conference room environment is constructed.

8. A delay calculation device based on a complex reverberation conference room environment, characterized in that: including a processor and a memory; The processor is connected to the memory; The memory is used to store executable program code; The processor runs a program corresponding to the executable program code by reading the executable program code stored in the memory, so as to execute the steps of the method according to any one of claims 1 to 6.

9. A computer-readable storage medium having a computer program stored thereon, characterized in that: The computer-readable storage medium stores instructions, and when the instructions are executed on a computer or a processor, the computer or the processor executes the steps of the method according to any one of claims 1 to 6.

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