WO2021057214A1

WO2021057214A1 - Sound field extension method, computer apparatus, and computer readable storage medium

Info

Publication number: WO2021057214A1
Application number: PCT/CN2020/102899
Authority: WO
Inventors: 戚炎兴
Original assignee: 深圳Tcl新技术有限公司
Priority date: 2019-09-29
Filing date: 2020-07-20
Publication date: 2021-04-01
Also published as: CN112584275A; CN112584275B

Abstract

The present disclosure relates to a sound field extension method, a computer apparatus, and a computer readable storage medium. The method comprises the following steps: acquiring at least two sound channel signals, wherein the at least two sound channel signals comprise a first sound channel signal and a second sound channel signal; acquiring a first gain signal and a third gain signal corresponding to the first sound channel signal, and a second gain signal corresponding to the second sound channel signal; obtaining a first loudness compensation signal according to the first gain signal and the second gain signal; and obtaining an output signal of the first sound channel signal according to the third gain signal and the first loudness compensation signal. The invention removes high-intensity sound by means of loudness compensation, and amplifies weak sounds, undetectable by human ears due to a masking effect, so that the weak sounds are perceptible for human ears, thereby improving the three-dimensional sound effect, and restoring the authentic sound field.

Description

Sound field expansion method, computer equipment and computer readable storage medium

priority

This disclosure requires the priority of a Chinese patent application filed with the Chinese Patent Office on September 29, 2019, the application number is "2019109363728", and the application name is "a sound field expansion method, computer equipment, and computer-readable storage medium" , The entire contents of which are incorporated into the present disclosure by reference.

Technical field

The present disclosure relates to the field of sound processing technology, and in particular, to a sound field expansion method, computer equipment, and computer-readable storage medium.

Background technique

Televisions and cinemas are audio-visual products, and picture quality and sound quality have always been consumers' concerns. In terms of sound, in addition to the quality of output sound, the sense of presence is also an aspect that users pay more and more attention to. The sound system of a TV is generally dual-channel, which can only reproduce part of the spatial information of the sound, and the sound heard by the user feels that it is transmitted from the TV, and cannot reproduce the three-dimensional spatial effect. Multi-channel playback systems are usually used to enhance the three-dimensional spatial effect, but special program sources and multi-speaker playback conditions are required, which limits the application of multi-channel playback systems. In the prior art, it is difficult to restore the three-dimensional spatial information of a sound channel with fewer sound channels.

Therefore, the existing technology needs to be improved.

Public content

The technical problem to be solved by the present disclosure is to provide a sound field expansion method, a computer device, and a computer-readable storage medium to realize the restoration of the three-dimensional spatial information of the sound channel.

On the one hand, an embodiment of the present disclosure provides a sound field expansion method, including the following steps:

Acquiring at least two channel signals, where the at least two channel signals include a first channel signal and a second channel signal;

Acquiring a first gain signal and a third gain signal corresponding to the first channel signal, and a second gain signal corresponding to the second channel signal;

To a first loudness compensation signal according to the first gain signal and the second gain signal;

Obtain the output signal of the first channel signal according to the third gain signal and the first loudness compensation signal.

In an implementation manner, the gain coefficient corresponding to the first gain signal is the same as the gain coefficient corresponding to the second gain signal, and the gain coefficient corresponding to the first gain signal is the same as the gain coefficient corresponding to the third gain signal. The sum of the coefficients is equal to the preset value;

The acquiring the first gain signal and the third gain signal corresponding to the first channel signal, and the second gain signal corresponding to the second channel signal includes:

Performing first gain processing and second gain processing on the first channel signal respectively to obtain the first gain signal and the third gain signal;

Performing first gain processing on the second channel signal to obtain the second gain signal.

In an implementation manner, the obtaining the first loudness compensation signal according to the first gain signal and the second gain signal includes:

Obtaining a first difference signal according to the first gain signal and the second gain signal;

Perform loudness compensation on the first difference signal to obtain the first loudness compensation signal.

In an implementation manner, the obtaining a first difference signal according to the first gain signal and the second gain signal includes:

Performing phase compensation on the second gain signal to obtain a first phase compensation signal;

The first difference signal is obtained according to the first gain signal and the first phase compensation signal.

In an implementation manner, the performing phase compensation on the second gain signal to obtain the first phase compensation signal includes:

Performing phase compensation on the second gain signal through the first digital filter bank to obtain the first phase compensation signal.

In an implementation manner, the first digital filter bank includes several digital filters, and the digital filter bank is expressed by the following formula:

Y=y ₁ (n)+y ₂ (n)+···+y _m (n)

Where, Y is the output sequence of the first digital filter bank, sampling point n≥0, and y _m (n) is the output sequence of the m-th digital filter.

In an implementation manner, the performing loudness compensation on the first difference signal to obtain the first loudness compensation signal includes:

Obtain a preset loudness compensation curve, and adjust the decibel value of the first difference signal according to the preset loudness compensation curve to obtain the first loudness compensation signal.

In an implementation manner, the preset loudness compensation curve is an equal loudness curve corresponding to the geographic area to which the user belongs.

In an implementation manner, the acquiring at least two channel signals, where after the at least two channel signals include a first channel signal and a second channel signal, further includes:

Acquiring a fourth gain signal corresponding to the second channel signal;

Obtaining a second loudness compensation signal according to the second gain signal and the first gain signal;

Adding the fourth gain signal and the second loudness compensation signal to obtain an output signal of the second channel signal.

In an implementation manner, the acquiring a fourth gain signal corresponding to the second channel signal includes:

Obtain the second channel signal, and perform second gain processing on the second channel signal to obtain the fourth gain signal.

In an implementation manner, the obtaining a second loudness compensation signal according to the second gain signal and the first gain signal includes:

Obtaining a second difference signal according to the second gain signal and the first gain signal;

Performing loudness compensation on the second difference signal to obtain the second loudness compensation signal.

In an implementation manner, the obtaining a second difference signal according to the second gain signal and the first gain signal includes:

Performing phase compensation on the first gain signal to obtain a second phase compensation signal;

Subtracting the second phase compensation signal from the second gain signal to obtain the second difference signal.

In an implementation manner, the performing phase compensation on the first gain signal to obtain the second phase compensation signal includes:

Performing phase compensation on the first gain signal through a third digital filter bank to obtain the second phase compensation signal.

In an implementation manner, the third digital filter bank includes several digital filters, and the digital filters are expressed by the following formula:

Y=y ₁ (n)+y ₂ (n)+···+y _m (n)

Wherein, Y is the output sequence of the third digital filter bank, sampling point n≥0, and y _m (n) is the output sequence of the m-th digital filter.

In an implementation manner, the performing loudness compensation on the second difference signal to obtain the second loudness compensation signal includes:

Obtain a preset loudness compensation curve, adjust the decibel value of the second difference signal according to the preset loudness compensation curve to obtain the second loudness compensation signal, and the preset loudness compensation curve is the equal loudness corresponding to the geographic area to which the user belongs curve.

In a second aspect, embodiments of the present disclosure provide a computer device including a memory and a processor, the memory stores a computer program, and the processor implements the following steps when the computer program is executed:

Obtaining a first loudness compensation signal according to the first gain signal and the second gain signal;

In a third aspect, embodiments of the present disclosure provide a computer-readable storage medium on which a computer program is stored, wherein the computer program implements the following steps when executed by a processor:

Compared with the prior art, the embodiments of the present disclosure have the following advantages:

According to the sound field expansion method provided by the embodiments of the present disclosure, at least two channel signals are acquired, where the at least two channel signals include a first channel signal and a second channel signal, and the signal corresponding to the first channel signal is acquired. According to the first gain signal and the third gain signal, and the second gain signal corresponding to the second channel signal, a first loudness compensation signal is obtained according to the first gain signal and the second gain signal; The three-gain signal and the first loudness compensation signal obtain an output signal of the first channel signal. When the sound field is expanded, this method removes the common component (stronger sound) in the first gain signal and the second gain signal, and then performs loudness compensation to amplify and enhance the weaker sound so that it is not Submerged by stronger sounds, and thus perceived by the human ear, these environmental information that is imperceptible to the human ear due to the masking effect can be perceived by the human ear, which improves the three-dimensional sound effect and restores the real sound field.

Description of the drawings

In order to more clearly illustrate the technical solutions in the embodiments of the present disclosure or the prior art, the following will briefly introduce the drawings that need to be used in the description of the embodiments or the prior art. Obviously, the drawings in the following description are only These are some embodiments described in the present disclosure. For those of ordinary skill in the art, other drawings can be obtained based on these drawings without creative work.

FIG. 1 is a schematic diagram of the position of the sound field expansion module in the system in an embodiment of the disclosure;

2 is a schematic flowchart of a sound field expansion method in an embodiment of the disclosure;

Fig. 3 is a phase diagram of a digital filter in an embodiment of the disclosure;

4 is an internal structure diagram of a sound field expansion processing module in an embodiment of the disclosure;

FIG. 5 is a curve diagram of medium loudness according to an embodiment of the disclosure;

6 is a schematic diagram of a preset loudness compensation curve designed for Asians in an embodiment of the disclosure;

Fig. 7 is an internal structure diagram of a computer device in an embodiment of the disclosure.

FIG. 8 is a flowchart of sound initialization in an embodiment of the disclosure.

detailed description

In order to enable those skilled in the art to better understand the solutions of the present disclosure, the technical solutions in the embodiments of the present disclosure will be described clearly and completely with reference to the accompanying drawings in the embodiments of the present disclosure. Obviously, the described embodiments are only These are a part of the embodiments of the present disclosure, but not all of the embodiments. Based on the embodiments in the present disclosure, all other embodiments obtained by those of ordinary skill in the art without creative work shall fall within the protection scope of the present disclosure.

After research, the inventor found that in order to improve the three-dimensional sound effect of the sound field, a two-channel or multi-channel system is usually used to enhance the three-dimensional sound effect of the sound field. However, there is a masking effect in the sound field, that is, the auditory perception of a weaker sound is changed by another. The phenomenon of the influence of stronger sound makes the human ears only hear the stronger sound but not the weaker sound, that is to say, the three-dimensional sound effect of the sound field is still poor.

In order to solve the above problems, in the embodiments of the present disclosure, the first gain signal and the second gain signal are obtained after gain processing is performed on the two channel signals respectively, and then the first gain signal and the second gain signal are obtained according to the first gain signal and the second gain signal. A difference signal, where the first difference signal does not include the sound shared by the two channel signals (that is, the stronger sound), but includes the weaker sound. Then the loudness compensation of the first difference signal will make the weaker sound The sound is amplified to obtain the first loudness compensation signal, and finally the original channel signal is compensated by the first loudness compensation signal (specifically, the third gain signal of the first channel signal subjected to the second gain processing) to obtain the first channel signal corresponding In the output signal at this time, the original weaker sound is enhanced without being masked by the stronger sound. The human ear can hear more sound components, and the three-dimensional sound effect is better.

For example, the embodiments of the present disclosure can be applied to the scenario shown in FIG. 1. The method of the present disclosure belongs to a sound post-processing procedure, and a sound field expansion processing module can be added to the existing sound processing procedure to realize the processing of digital sound signals. In the TV sound processing process shown in FIG. 1, the sound field The expansion processing module is located after all existing sound processing procedures and before output to the power amplifier.

Specifically, in this scenario, first, the analog audio signal and the digitally encoded audio signal are respectively A/D converted and audio decoded to obtain the digital audio signal, and then the digital audio signal is subjected to EQ processing, noise reduction and other processing, and then Sound field expansion. Specifically, the sound field expansion adopts the sound field expansion method in this embodiment to process the digital sound signal to obtain the output signal. The final output signal can be output through an analog power amplifier or headphones after D/A conversion, or through a digital power amplifier after I2S conversion.

It should be pointed out that the sound field expansion method can be embedded in the existing sound processing flow, or a separate sound expansion module can be formed to process existing sounds.

It should be noted that the above application scenarios are only shown to facilitate the understanding of the present disclosure, and the embodiments of the present disclosure are not limited in this respect. On the contrary, the embodiments of the present disclosure can be applied to any applicable scenarios.

Hereinafter, various non-limiting embodiments of the present disclosure will be described in detail with reference to the accompanying drawings.

Referring to Fig. 2, there is shown a sound field expansion method in an embodiment of the present disclosure. In this embodiment, the method may include the following steps, for example:

S1. Obtain at least two channel signals, where the at least two channel signals include a first channel signal and a second channel signal.

In the embodiment of the present disclosure, the first channel signal and the second channel signal are from different channels, that is, the method described in the present disclosure can be applied to a dual-channel system or a multi-channel system. In order to achieve three-dimensional sound effects and create a sound field that tends to be realistic, in a two-channel system or a multi-channel system, two or two channels play different channel signals. Then, the first channel signal and the second channel signal The two-channel signal is not exactly the same.

Generally, some televisions, mobile terminals, and other devices that implement sound playback through power amplifiers or earphones use a two-channel system. When the method of the present disclosure is applied to a two-channel system, for example, the left channel is used for both channels. And right channel, the first channel information comes from the left channel, and the second channel information comes from the right channel.

Usually, some commercial theaters, home theaters, and other large-scale sound playback devices use multi-channel systems. The multi-channel system here refers to a system composed of three or more channels. This disclosure When the method is applied to a multi-channel system, for example, four channels are used, including: a front left channel, a front right channel, a rear left channel, and a rear right channel. The second channel signal can come from any two of the four channels. It should be noted that the three-dimensional sound effect can be improved by increasing the number of channels. However, in large-scale situations, the number of channels cannot be increased without limitation. Increasing the number of channels will also bring about exponentially multiplied sound processing difficulties. The disclosed method can be combined with increasing the number of channels to improve the three-dimensional sound effect and construct a real sound field.

In a two-channel system or a multi-channel system, only any one or more of the channels can be subjected to sound field expansion. For example, in an upper, lower, left, and right four-channel system, only the upper channel signal may be subjected to sound field expansion processing, or only the upper channel signal and the left channel signal may be subjected to sound field expansion processing.

In a multi-channel system, the first channel signal and the second channel signal can be any two channel signals, and after performing sound field expansion processing on the first channel signal based on the second channel signal, it can also be based on The remaining channel signals perform sound field expansion processing on the first channel signal.

For example, in a four-channel system of upper, lower, left, and right, the upper channel signal can be expanded according to the lower channel signal, and then the upper channel signal can be continued according to the left channel signal or the right channel signal. The channel signal undergoes sound field expansion processing. That is, first the lower channel signal and the upper channel signal obtain the corresponding output signal, and the other output signal is obtained from the output signal and the left channel signal (or the right channel signal).

The method described in the present disclosure is the calculation and processing of time domain signals. The first channel signal and the second channel signal belong to time domain signals, and their signals change with time. Specifically, the analog and digital sound sources input by the TV are uniformly processed into a normalized digital audio signal (for example, two-channel PCM, 48000 Hz sampling rate, 16 bits). After the digital audio signal undergoes EQ processing, noise reduction and other processing, the input two-channel PCM data is processed by the sound field expansion processing module, and then the processed two-channel PCM data is output to the back-end module.

The following implementation takes a two-channel system as an example for description:

To facilitate understanding, the first channel is defined as the left channel in a two-channel system, and the second channel is defined as the right channel in a two-channel system. The first channel signal is the left channel signal, and the second channel is defined as the right channel in the two-channel system. The two-channel signal is the right-channel signal. It should be noted that in another implementation manner, the first channel can be defined as the right channel in a two-channel system, and the second channel can be defined as the left channel in a two-channel system. In other channel channel systems, the first channel and the second channel can be any two channels. In this embodiment, it is not limited which channel the first channel and the second channel are specifically.

S2. Acquire a first gain signal and a third gain signal corresponding to the first channel signal, and a second gain signal corresponding to the second channel signal.

Since the output signal of the first channel signal is obtained based on the first channel signal and the second channel signal, in order to prevent sound distortion, it is necessary to compare the first channel signal and the second channel signal. The channel signal undergoes gain processing. The corresponding gain processing of the first gain signal and the third gain signal may be the same or different. For example, the first gain signal is 0.5 times the first channel signal, and the third gain signal The gain signal is 0.5 times the first channel signal; or the first gain signal is 0.25 times the first channel signal, and the third gain signal is 0.7 times the first channel signal . The gain processing of the first gain signal and the second gain signal may be the same or different. For example, the first gain signal is 0.25 times the first channel signal, and the second gain signal 0.25 times the second channel signal; or the first gain signal is 0.5 times the first channel signal, and the second gain signal is 0.45 times the second channel signal.

In this embodiment, the gain coefficient corresponding to the first gain signal is the same as the gain coefficient corresponding to the second gain signal, and the gain coefficient corresponding to the first gain signal is denoted as b, then the second gain signal corresponds to The gain factor of is also b. The sum of the gain coefficient b corresponding to the first gain signal and the gain coefficient a corresponding to the third gain signal is equal to a preset value, that is, a+b=the preset value.

Specifically, S2 includes:

S21: Perform first gain processing and second gain processing on the first channel signal respectively to obtain a first gain signal and a third gain signal.

S22. Perform first gain processing on the second channel signal to obtain a second gain signal.

After acquiring the first channel signal and the second channel signal, perform first gain processing on the first channel signal, perform first gain processing on the second channel signal, and perform the first gain processing on the second channel signal. The gain processing is b times gain processing. The first channel signal is denoted as Lin, and the second channel signal is denoted as Rin. After b-fold gain processing is performed on the first channel signal, the first gain signal is obtained, denoted as b*Lin, and after b-fold gain processing is performed on the second channel signal, the second gain is obtained Signal, denoted as b*Rin, b=0.2-0.3.

After acquiring the first channel signal, perform second gain processing on the first channel signal to obtain a third gain signal, the second gain processing is a multiplier gain processing, and the third gain signal is denoted as a*Lin, a=0.7-0.8. In order to prevent sound distortion, a and b can be adjusted according to needs, and need to satisfy 0<a<1, 0<b<1, and a+b≈1. For example, a+b=1±0.1 can be limited, that is, pre- Set the value to 0.9-1.1.

Of course, in another implementation manner, the second gain processing may be performed on the second channel signal. Which channel signal is subjected to the second gain processing, and the finally obtained output signal is used as the corresponding output signal of the channel signal. Here, the first channel signal is taken as an example for description, that is, the final output signal Is the output signal as the first channel signal.

S3. Obtain a first loudness compensation signal according to the first gain signal and the second gain signal.

After performing gain processing on the first channel signal and the second channel signal, the same components in the first gain signal and the second gain signal are removed to obtain the first gain signal and the Different components in the second gain signal are then subjected to loudness compensation to obtain the first loudness compensation signal.

Specifically, S3 includes:

S31. Obtain a first difference signal according to the first gain signal and the second gain signal.

Specifically, the first difference signal can be obtained by subtracting two or two channel signals after the first gain processing. Since the first difference signal has a sign difference, the first gain signal and the second gain signal are both It can be used as a subtracted number or a subtracted number. When the final output signal is used as the output signal of the first channel signal, the first gain signal is used to subtract the second gain signal; when the final output signal is When the signal is used as the output signal of the second channel signal, the second gain signal is used to subtract the first gain signal. As mentioned above, the final output signal is taken as an example of the output signal of the first channel signal. Therefore, the first difference signal is obtained according to the first gain signal and the second gain signal, Specifically, it means that the first gain signal is subtracted from the second gain signal to obtain the first difference signal.

Since the phase of the acoustic wave signal will change after the loudness compensation process, it is necessary to perform phase compensation on the acoustic wave signal. The phase compensation process can be placed before the loudness compensation process or after the loudness compensation process. When the phase compensation process is placed before the loudness compensation process, it has a better compensation effect. Therefore, the phase compensation process is placed on the loudness in this disclosure. Before compensation processing.

The phase compensation processing can be phase compensation for the first difference signal, phase compensation for the first gain signal or the second gain signal, or phase compensation for the first channel signal or the second channel signal. Since the first gain signal is used to remove the same component in the second gain signal, the present disclosure only performs phase compensation on the second gain signal.

Specifically, step S31 includes:

S311: Perform phase compensation on the second gain signal to obtain a first phase compensation signal.

Specifically, the adjustment range of the phase compensation is 0-π. It should be pointed out that the adjustment range of phase compensation adjustment (ie phase compensation value) is adjusted according to the frequency band, and the amplitude of adjustment of different frequency bands is different. That is to say, the phase compensation value of some frequency bands is small and can be lower. To 0, the phase compensation value of some frequency bands is higher, which can be as high as π.

The phase compensation value is related to the frequency band of the sound. In the low frequency band, the phase compensation value can be 0-π/9; in the middle frequency band, the phase compensation value can be π/9-π/3; in the high frequency band, the phase compensation value can be used. The value can be π/3-π.

After the loudness compensation process (in this embodiment, a digital filter is used for loudness compensation), the phase of the acoustic signal will change. As shown in Figure 4, after loudness compensation, the signal below 400Hz has a phase lead, while the phase in the 400～1.5KHz interval lags slightly, and the phase in the frequency band above 1.5KHz leads. This phase distortion is not required by the embodiment of the present disclosure, and the phase of the acoustic signal needs to be corrected. In the mid-high frequency range (above 1.5KHz), the phase lead affects the sound field broadening effect. The phase compensation filter needs to delay the phase in the mid-high frequency range (above 1.5KHz) and the phase in the low frequency range (below 400Hz) at the same time.

Because the phase of the acoustic signal will change, as shown in Figure 3, the phase compensation is performed before the loudness compensation, so as to pre-correct the phase distortion caused by the subsequent loudness compensation. Of course, the phase compensation also affects the acoustic signal (specifically, the second Gain signal) to perform phase compensation processing. Perform phase compensation on the second gain signal to obtain a first phase compensation signal, denoted as H(b*Rin), and H(·) represents the transfer function of the phase compensation.

When phase compensation is performed on the acoustic signal, corresponding phase compensation is performed on the acoustic signal of different frequencies. At the same time, it is necessary to ensure that the amplitude (amplitude) of the acoustic signal does not produce a large change. In order to reduce the amount of calculation, the embodiment of the present disclosure performs phase compensation on the second gain signal through the first digital filter bank to obtain the first phase compensation signal.

The first digital filter bank includes several digital filters, and is expressed by the following formula:

Y=y ₁ (n)+y ₂ (n)+···+y _m (n)

Where, Y is the output sequence of the first digital filter bank, sampling point n≥0, and y _m (n) is the output sequence of the m-th digital filter. The several digital filters form a direct structure.

According to the time domain characteristics of the impulse response function, the digital filter can be divided into two types, namely, an infinite impulse response (IIR) digital filter and a finite impulse response (FIR) digital filter.

When the digital filter adopts an IIR digital filter, the digital filter is expressed by the following formula (differential equation):

Where n is the sampling point, y _m (n) is the output sequence of the m-th digital filter, x _m (n) is the input sequence of the m-th digital filter, and a _i and b _i are all The coefficients of the digital filter, N is the order of the digital filter, i=0, 1, 2, ..., N, and Σ are the summation signs.

When the digital filter adopts the FIR digital filter, the digital filter adopts the following formula (differential equation) to express:

Where n is the sampling point, y _m (n) is the output sequence of the m-th digital filter, x _m (n) is the input sequence of the m-th digital filter, and b _i is the digital filter The coefficient of the filter, N is the order of the digital filter, i=0, 1, 2, ···, N, Σ is the summation sign.

Through the coefficients a _i and b _{i of} the digital filter, it is ensured that the amplitude of the sound in the frequency range (specifically 20-20 KHz) that can be heard by the human ear does not produce a large change. When the FIR digital filter is used, the order of the filter is required to be higher to meet the requirements. Therefore, the calculation amount of the FIR digital filter is larger, and the calculation amount of the IIR digital filter is smaller.

S312. Subtract the first phase compensation signal from the first gain signal to obtain a first difference signal.

The first gain signal is subtracted from the first phase compensation signal to obtain the first difference signal. The first difference signal obtained here is used to obtain the output signal corresponding to the first channel signal. A difference signal is denoted as b*Lin-H(b*Rin).

S32. Perform loudness compensation on the first difference signal to obtain a first loudness compensation signal.

Specifically, loudness compensation is performed on the first difference signal to obtain the first loudness compensation signal, which is denoted as Q(b*Lin-H(b*Rin)), and Q(·) represents the transfer function of loudness compensation. The first gain signal and the first phase compensation signal are not exactly the same. After the subtraction process is performed, the common components of the two are removed, and the difference between the two is retained in the first difference signal. The same components, for example, the first difference signal includes environmental information during sound recording, and these environmental signals are amplified after loudness compensation.

Specifically, when the human ear receives two or more sounds, there is a masking effect, that is, the auditory perception of a weaker sound is affected by another stronger sound. In a quiet environment, the ear can distinguish the slight sound, but in a noisy environment, the slight sound will be completely submerged. If you want to hear the original slight sound, you must increase the slight sound. The method of the present disclosure removes the stronger sound (that is, the component common to both) by making the difference between the first gain signal and the first phase compensation signal, and then performs loudness compensation to amplify and enhance the weaker sound , So that it will not be submerged by strong sounds, and thus be perceived by the human ear. These environmental information that is imperceptible to the human ear due to the masking effect can be perceived by the human ear, which improves the three-dimensional sound effect and restores the real sound field. Real three-dimensional spatial information.

Specifically, a preset loudness compensation curve is acquired, and the decibel value of the first difference signal is adjusted according to the preset loudness compensation curve to obtain the first loudness compensation signal.

After subtracting the first phase compensation signal from the first gain signal to obtain the first difference signal, the preset loudness compensation curve is obtained, and the preset loudness compensation curve is the equal loudness curve corresponding to the geographic area to which the user belongs Of course, the preset loudness compensation curve may also be a curve obtained by adjusting according to actual sound effects and based on the equal loudness curve corresponding to the geographic area to which the user belongs. The equal loudness curve refers to the curve of the relationship between the sound pressure level and the frequency of a pure tone with the same loudness perceived by a typical listener. When external sounds are introduced into human ears, people form the concept of auditory sound strength in a subjective sense, and use "ring" and "not ringing" to describe the strength of the sound. In addition to the amplitude of the sound wave, the human ear’s perception of sound loudness is also related to the frequency of the sound wave. Even sounds with the same sound pressure level but different frequencies will sound different to the human ear. The equal loudness curve shown in Figure 5 indicates the sound pressure level requirements for sounds in different frequency bands when the human ear feels the same loudness (that is, volume) in different frequency bands under different sound pressure levels. . For example, the bottom solid line indicates that when the human ear feels the same loudness, the sound pressure level of 100Hz sound needs about 30dB, while the sound pressure level of 1K-2KHz sound only needs to reach about 10dB. For 4KHz sound, the sound pressure level is as low as less than 5dB.

The equal-loudness curve is a statistical curve that takes into account the auditory characteristics of the population, that is, people in different geographic regions have corresponding equal-loudness curves. Asians and Europeans have different equal-loudness curves, so it needs to be based on users. Select the corresponding equal loudness curve for the geographic area. Figure 6 shows the preset loudness compensation curve designed for Asians.

In the equal loudness curve, a series of curves are formed according to the magnitude of the loudness. When obtaining the preset loudness compensation curve, it is necessary to obtain the corresponding preset loudness compensation curve according to the user's instruction to set the volume parameter and the effect parameter. The configuration file of the effect parameter is stored in the ini format, and for the convenience of program call, as shown in Figure 8, during the boot initialization stage, the content of the parameter configuration file of the project configuration needs to be parsed and transferred to the module through the interface provided by the module. The sound field expansion module then selects the corresponding parameters according to the interval where the current volume value of the system is located.

In the embodiment of the present disclosure, the second digital filter bank is used to perform loudness compensation, and the decibel value of the first difference signal is adjusted so that the decibel value of the first difference signal is close to the data point of the preset loudness compensation curve. The first loudness compensation signal is denoted as Q(b*Lin-H(b*Rin)). The abscissa of the data point of the preset loudness compensation curve is the frequency, and the ordinate is the sound pressure level (that is, the decibel value), and the decibel value of the first difference signal is adjusted with respect to the frequency of the first difference signal, The decibel value of the first difference signal tends to the decibel value of the corresponding data point at the same frequency on the preset loudness compensation curve.

The second digital filter bank includes several digital filters, and is expressed by the following formula:

Y=y ₁ (n)+y ₂ (n)+···+y _m (n)

Where, Y is the output sequence of the second digital filter bank, sampling point n≥0, and y _m (n) is the output sequence of the m-th digital filter. The several digital filters form a direct structure.

Where n is the sampling point, y _m (n) is the output sequence of the m-th digital filter, x _m (n) is the input sequence of the m-th digital filter, and a _i and b _i are all For the coefficient of the digital filter, N is the order of the digital filter; i=0, 1, 2, ···, N; Σ is the sum sign.

The difference from the first digital filter bank is that the coefficients a _i and b _i of the digital filters in the second digital filter bank are different, and the order N is different. When the FIR digital filter is used, the order of the filter is required to be higher to meet the requirements. Therefore, the calculation amount of the FIR digital filter is larger, and the calculation amount of the IIR digital filter is smaller.

S4. Obtain an output signal of the first channel signal according to the third gain signal and the first loudness compensation signal.

Specifically, the third gain and the first loudness compensation signal are added to obtain the output signal of the first channel signal.

In the embodiment of the present disclosure, the first loudness compensation signal (ie Q(b*Lin-H(b*Rin))) is added to the third gain signal to obtain the output signal of the first channel signal, so The output signal of the first channel signal is denoted as Lout, then Lout=a*Lin+Q(b*Lin-H(b*Rin)). The output signal of the first channel signal contains environmental information that is submerged in the sound due to the masking effect, which improves the three-dimensional sound effect, expands the sound field, and enhances the sense of presence. In addition, the method described in the present disclosure is only implemented by software algorithms, and no additional hardware cost is required.

The above describes the process of obtaining the output signal of the first channel signal, which is equivalent to replacing the original first channel signal with the output signal of the first channel signal. The following describes the use of similar steps to obtain the output signal of the second channel signal .

After the S1, it also includes:

S5. Obtain a fourth gain signal corresponding to the second channel signal.

Specifically, the second channel signal is acquired, and the second gain processing is performed on the second channel signal to obtain a fourth gain signal.

Specifically, the second gain processing is performed on the second channel signal to obtain a fourth gain signal, the second gain processing is a multiplication gain processing, and the fourth gain signal is denoted as a*Rin, a=0.5- 0.9.

S6. Obtain a second loudness compensation signal according to the second gain signal and the first gain signal.

Specifically, S6 includes:

S61. Obtain a second difference signal according to the second gain signal and the first gain signal.

Specifically, S61 includes:

S611: Perform phase compensation on the first gain signal to obtain a second phase compensation signal.

S612. Subtract the second phase compensation signal from the second gain signal to obtain a second difference signal.

Perform phase compensation on the first gain signal to obtain a second phase compensation signal, denoted as H(b*Lin), and H(·) represents the transfer function of the phase compensation.

The second gain signal is subtracted from the second phase compensation signal to obtain the second difference signal, and the second difference signal is denoted as b*Rin-H(b*Lin).

Then loudness compensation is performed on the second difference signal to obtain the second loudness compensation signal, denoted as Q(b*Rin-H(b*Lin)), and Q(·) represents the transfer function of the loudness compensation.

Specifically, the second phase compensation signal is obtained by performing phase compensation on the first gain signal through the third digital filter bank.

The third digital filter bank includes several digital filters, and is expressed by the following formula:

Y=y ₁ (n)+y ₂ (n)+···+y _m (n)

Wherein, Y is the output sequence of the third digital filter bank, sampling point n≥0, and y _m (n) is the output sequence of the m-th digital filter. The several digital filters form a direct structure.

S62. Perform loudness compensation on the second difference signal to obtain a second loudness compensation signal.

Specifically, a preset loudness compensation curve is obtained, and the decibel value of the second difference signal is adjusted according to the preset loudness compensation curve to obtain the second loudness compensation signal.

In the embodiment of the present disclosure, the fourth digital filter bank is used to perform loudness compensation, and the decibel value of the second difference signal is adjusted so that the decibel value of the second difference signal is close to the data point of the preset loudness compensation curve. The second loudness compensation signal is denoted as Q(b*Rin-H(b*Lin)). The abscissa of the data point of the preset loudness compensation curve is the frequency, and the ordinate is the sound pressure level (ie, decibel value), adjust the decibel value of the second difference signal with respect to the frequency of the second difference signal, The decibel value of the second difference signal tends to the decibel value of the data point corresponding to the same frequency on the preset loudness compensation curve.

The fourth digital filter bank includes several digital filters, and is expressed by the following formula:

Y=y ₁ (n)+y ₂ (n)+···+y _m (n)

Wherein Y is the output sequence of the fourth digital filter bank, sampling point n≥0, and y _m (n) is the output sequence of the m-th digital filter. The several digital filters form a direct structure.

The difference from the third digital filter bank is that the coefficients a _i and b _i of the digital filters in the fourth digital filter bank and the order N are different. When the FIR digital filter is used, the order of the filter is required to be higher to meet the requirements. Therefore, the calculation amount of the FIR digital filter is larger, and the calculation amount of the IIR digital filter is smaller.

S7. Adding the fourth gain signal and the second loudness compensation signal to obtain an output signal of the second channel signal.

In the embodiment of the present disclosure, the second loudness compensation signal (ie, Q(b*Rin-H(b*Lin))) is added to the fourth gain signal to obtain the output signal of the second channel signal. Rout, then Rout=a*Rin+Q(b*Rin-H(b*Lin)). The output signal of the second channel signal contains environmental information that is submerged in the sound due to the masking effect, which improves the three-dimensional sound effect, expands the sound field, and enhances the sense of presence. In addition, the method described in the present disclosure is only implemented by software algorithms, and no additional hardware cost is required.

By separately expanding the sound field of the two channels in the dual-channel system, the three-dimensional sound effects are further improved from the left and right channels, and the sense of presence is enhanced.

When the sound field is expanded, this method removes the common component (stronger sound) in the first gain signal and the second gain signal, and then performs loudness compensation to amplify and enhance the weaker sound so that it is not It is submerged by strong sound and thus is perceived by the human ear. These environmental information that is imperceptible to the human ear due to the masking effect can be perceived by the human ear, which improves the three-dimensional sound effect and restores the real sound field. Furthermore, the phase compensation is performed before the loudness compensation, so as to correct the phase distortion caused by the subsequent loudness compensation in advance.

In one embodiment, the present disclosure provides a computer device, which may be a terminal, and the internal structure is shown in FIG. 7. The computer equipment includes a processor, a memory, a network interface, a display screen and an input device connected through a system bus. Among them, the processor of the computer device is used to provide calculation and control capabilities. The memory of the computer device includes a non-volatile storage medium and an internal memory. The non-volatile storage medium stores an operating system and a computer program. The internal memory provides an environment for the operation of the operating system and computer programs in the non-volatile storage medium. The network interface of the computer device is used to communicate with an external terminal through a network connection. The computer program is executed by the processor to realize the sound field expansion method. The display screen of the computer equipment can be a liquid crystal display screen or an electronic ink display screen, and the input device of the computer equipment can be a touch layer covered on the display screen, or it can be a button, a trackball or a touchpad set on the housing of the computer equipment , It can also be an external keyboard, touchpad, or mouse.

Those skilled in the art can understand that the block diagram shown in FIG. 7 is only a partial structure related to the solution of the present disclosure, and does not constitute a limitation on the computer device to which the solution of the present disclosure is applied. A specific computer device may include a diagram. More or fewer parts are shown in, or some parts are combined, or have a different arrangement of parts.

In one embodiment, a computer device is provided, including a memory and a processor, the memory stores a computer program, and the processor implements the following steps when executing the computer program:

In one embodiment, a computer-readable storage medium is provided, on which a computer program is stored, and when the computer program is executed by a processor, the following steps are implemented:

The technical features of the above embodiments can be combined arbitrarily. In order to make the description concise, all possible combinations of the technical features in the above embodiments are not described. However, as long as there is no contradiction in the combination of these technical features, they should be It is considered as the range described in this specification.

Claims

A sound field expansion method, wherein the method includes the following steps:

Acquiring at least two channel signals, where the at least two channel signals include a first channel signal and a second channel signal;

Acquiring a first gain signal and a third gain signal corresponding to the first channel signal, and a second gain signal corresponding to the second channel signal;

Obtaining a first loudness compensation signal according to the first gain signal and the second gain signal;

Obtain the output signal of the first channel signal according to the third gain signal and the first loudness compensation signal.
The method according to claim 1, wherein the gain coefficient corresponding to the first gain signal is the same as the gain coefficient corresponding to the second gain signal, and the gain coefficient corresponding to the first gain signal is the same as the gain coefficient corresponding to the third gain signal. The sum of the gain coefficients corresponding to the signal is equal to the preset value;

The acquiring the first gain signal and the third gain signal corresponding to the first channel signal, and the second gain signal corresponding to the second channel signal includes:

Performing first gain processing and second gain processing on the first channel signal respectively to obtain the first gain signal and the third gain signal;

Performing first gain processing on the second channel signal to obtain the second gain signal.
The method according to claim 1, wherein the obtaining the first loudness compensation signal according to the first gain signal and the second gain signal comprises:

Obtaining a first difference signal according to the first gain signal and the second gain signal;

Perform loudness compensation on the first difference signal to obtain the first loudness compensation signal.
The method according to claim 3, wherein the obtaining the first difference signal according to the first gain signal and the second gain signal comprises:

Performing phase compensation on the second gain signal to obtain a first phase compensation signal;

The first difference signal is obtained according to the first gain signal and the first phase compensation signal.
The method according to claim 4, wherein the performing phase compensation on the second gain signal to obtain the first phase compensation signal comprises:

Performing phase compensation on the second gain signal through the first digital filter bank to obtain the first phase compensation signal.
The method according to claim 5, wherein the first digital filter bank includes a plurality of digital filters, and the digital filter bank is expressed by the following formula:

Y=y 1 (n)+y 2 (n)+…+y m (n)

Where, Y is the output sequence of the first digital filter bank, sampling point n≥0, and y m (n) is the output sequence of the m-th digital filter.
The method according to any one of claims 3-6, wherein the performing loudness compensation on the first difference signal to obtain the first loudness compensation signal comprises:

Obtain a preset loudness compensation curve, and adjust the decibel value of the first difference signal according to the preset loudness compensation curve to obtain the first loudness compensation signal.
8. The method according to claim 7, wherein the preset loudness compensation curve is an equal loudness curve corresponding to the geographic area to which the user belongs.
The method according to claim 1, wherein the acquiring at least two channel signals, wherein after the at least two channel signals include a first channel signal and a second channel signal, further comprising:

Acquiring a fourth gain signal corresponding to the second channel signal;

Obtaining a second loudness compensation signal according to the second gain signal and the first gain signal;

Adding the fourth gain signal and the second loudness compensation signal to obtain an output signal of the second channel signal.
The method according to claim 9, wherein said obtaining a fourth gain signal corresponding to said second channel signal comprises:

Obtain the second channel signal, and perform second gain processing on the second channel signal to obtain the fourth gain signal.
The method according to claim 9, wherein the obtaining a second loudness compensation signal according to the second gain signal and the first gain signal comprises:

Obtaining a second difference signal according to the second gain signal and the first gain signal;

Performing loudness compensation on the second difference signal to obtain the second loudness compensation signal.
The method according to claim 11, wherein said obtaining a second difference signal according to said second gain signal and said first gain signal comprises:

Performing phase compensation on the first gain signal to obtain a second phase compensation signal;

Subtracting the second phase compensation signal from the second gain signal to obtain the second difference signal.
The method according to claim 12, wherein said performing phase compensation on said first gain signal to obtain a second phase compensation signal comprises:

Performing phase compensation on the first gain signal through a third digital filter bank to obtain the second phase compensation signal.
The method according to claim 13, wherein the third digital filter bank includes a plurality of digital filters, and the digital filters are expressed by the following formula:

Y=y 1 (n)+y 2 (n)+…+y m (n)

Wherein, Y is the output sequence of the third digital filter bank, sampling point n≥0, and y m (n) is the output sequence of the m-th digital filter.
The method according to any one of claims 11-14, wherein the performing loudness compensation on the second difference signal to obtain the second loudness compensation signal comprises:

Obtain a preset loudness compensation curve, adjust the decibel value of the second difference signal according to the preset loudness compensation curve to obtain the second loudness compensation signal, and the preset loudness compensation curve is the equal loudness corresponding to the geographic area to which the user belongs curve.
A computer device comprising a memory and a processor, the memory storing a computer program, wherein the processor implements the steps of the method according to any one of claims 1 to 15 when the computer program is executed by the processor.
A computer-readable storage medium having a computer program stored thereon, wherein the computer program implements the steps of the method according to any one of claims 1 to 15 when the computer program is executed by a processor.