CN110719564B - Sound effect processing method and device - Google Patents
Sound effect processing method and device Download PDFInfo
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- CN110719564B CN110719564B CN201810772212.XA CN201810772212A CN110719564B CN 110719564 B CN110719564 B CN 110719564B CN 201810772212 A CN201810772212 A CN 201810772212A CN 110719564 B CN110719564 B CN 110719564B
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
- H04S7/30—Control circuits for electronic adaptation of the sound field
- H04S7/305—Electronic adaptation of stereophonic audio signals to reverberation of the listening space
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04S—STEREOPHONIC SYSTEMS
- H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
- H04S2420/01—Enhancing the perception of the sound image or of the spatial distribution using head related transfer functions [HRTF's] or equivalents thereof, e.g. interaural time difference [ITD] or interaural level difference [ILD]
Abstract
The invention provides a sound effect processing method and device. The method comprises the following steps: dividing each path of input sound signals into at least two paths of signals; filtering a first path of signal in the at least two paths of signals to obtain a first sound field signal; the coefficient of the filter adopted in the filtering processing is obtained by utilizing a self-adaptive algorithm according to the first sound field signal and the sound signals of the at least two sound channels, so that the correlation coefficient of the sound signals of the at least two sound channels and the first sound field signal is larger than a preset threshold value; delaying a second path of signals in the at least two paths of signals, and subtracting the first sound field signals from the delayed signals to obtain second sound field signals; and sending the first sound field signal to the central loudspeaker, and sending the second sound field signal to the corresponding loudspeaker. According to the embodiment of the invention, the sound signals of at least two sound channels are separated into the first sound field signal with strong correlation in the sound signals of at least two sound channels, so that the sound image quality of the middle sound field is improved.
Description
Technical Field
The invention relates to the technical field of audio, in particular to a sound effect processing method and device.
Background
The virtual surround sound technology is based on double-track stereo, without adding a track and a sound box, sound field signals are processed and played, so that a listener feels that the sound comes from a plurality of directions, and a simulated stereo sound field is generated. At present, the virtual surround technology is widely applied to televisions and virtual reality equipment.
In general, listening to a stereo signal at the vertices of an equilateral triangle formed by a stereo speaker and a listener can obtain a high-quality stereo sound image. Since the positional relationship between the user and the two speakers is not necessarily an equilateral triangle, the quality of the middle position of the sound image is deteriorated. These quality degradations include difficulty listening to the conversation, blurry sounds, unnatural speech sounds spoken, and the like. To address this problem, multiple channels may be employed, such as 5.1 or 7.1 channels, with a special intermediate channel added to the center speaker. The signal output to the center speaker is extracted from the stereo signal by signal processing.
In the related art, signals of left and right channels in a stereo signal may be added up and divided by 2, and the resultant signals may be output to a center speaker. However, the left and right channel signals inherently contain position information of the left and right sides, and thus the sound localization quality is poor after simple processing.
Disclosure of Invention
The embodiment of the invention provides a sound effect processing method and device, aiming at solving the problem that the quality of sound images is reduced due to the position change of a user relative to left and right loudspeakers.
In a first aspect, the present invention provides a sound effect processing method, including:
dividing each path of input sound signals into at least two paths of signals; the sound signals comprise sound signals of at least two sound channels;
filtering a first path of signal in the at least two paths of signals to obtain a first sound field signal; the coefficients of the filter adopted in the filtering process are obtained by using an adaptive algorithm according to the first sound field signal and the sound signals of the at least two sound channels, so that the correlation coefficients of the sound signals of the at least two sound channels and the first sound field signal are greater than a preset threshold;
delaying a second path of signal in the at least two paths of signals, and subtracting the first sound field signal from the delayed signal to obtain a second sound field signal;
and sending the first sound field signal to a central loudspeaker, and sending the second sound field signal to a corresponding loudspeaker.
In a second aspect, the present invention provides an audio processing apparatus, comprising:
the preprocessing module is used for dividing each path of input sound signals into at least two paths of signals; the sound signals comprise sound signals of at least two sound channels;
the processing module is used for filtering a first path of signal in the at least two paths of signals to obtain a first sound field signal; the coefficients of the filter adopted in the filtering process are obtained by using an adaptive algorithm according to the first sound field signal and the sound signals of the at least two sound channels, so that the correlation coefficients of the sound signals of the at least two sound channels and the first sound field signal are greater than a preset threshold;
the processing module is further configured to perform delay processing on a second path of signals in the at least two paths of signals, and subtract the first sound field signal from the delayed signals to obtain a second sound field signal;
and the sending module is used for sending the first sound field signal to a central loudspeaker and sending the second sound field signal to a corresponding loudspeaker.
According to the sound effect processing method and device provided by the embodiment of the invention, each input sound signal is divided into at least two paths of signals; filtering a first path of signal in the at least two paths of signals to obtain a first sound field signal; the coefficients of the filter adopted in the filtering process are obtained by using an adaptive algorithm according to the first sound field signal and the sound signals of the at least two sound channels, so that the correlation coefficients of the sound signals of the at least two sound channels and the first sound field signal are greater than a preset threshold; delaying a second path of signal in the at least two paths of signals, and subtracting the first sound field signal from the delayed signal to obtain a second sound field signal; the first sound field signal is sent to the central loudspeaker, the second sound field signal is sent to the corresponding loudspeaker, the first sound field signal and the second sound field signal are separated from the input sound signal, the first sound field signal is sent to the central loudspeaker, the second sound field signal is sent to the corresponding loudspeaker, the first sound field signal and the second sound field signal can be processed independently, the first sound field signal is a part with strong correlation in the sound signals of at least two sound channels, and the sound image quality of a middle sound field is improved.
Drawings
The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate embodiments consistent with the present disclosure and together with the description, serve to explain the principles of the disclosure.
FIG. 1 is a schematic view of a stereo image provided by the present invention;
FIG. 2 is a schematic view of a stereo image provided by the present invention;
FIG. 3 is a schematic diagram of a stereo image provided by the present invention;
FIG. 4 is a schematic diagram of an embodiment of a method provided by the present invention;
FIG. 5 is a schematic diagram of the distribution of musical instruments according to an embodiment of the method provided by the present invention;
FIG. 6 is a flowchart illustrating a sound processing method according to an embodiment of the present invention;
FIG. 7 is a schematic diagram illustrating a first embodiment of a sound effect processing method according to the present invention;
FIG. 8 is a schematic diagram of a second embodiment of a sound effect processing method according to the present invention;
FIG. 9 is a schematic diagram of a third embodiment of a sound effect processing method according to the present invention;
FIG. 10 is a schematic diagram illustrating a fourth embodiment of a sound effect processing method according to the present invention;
FIG. 11 is a schematic diagram illustrating a fifth embodiment of a sound effect processing method according to the present invention;
FIG. 12 is a schematic diagram of processing an intermediate sound field signal according to an embodiment of the sound effect processing method of the present invention;
FIG. 13 is a schematic diagram of an audio processing method according to another embodiment of the present invention;
FIG. 14 is a schematic diagram of processing an intermediate sound field signal according to another embodiment of the sound effect processing method of the present invention;
FIG. 15 is a schematic diagram of other sound field signal processing according to an embodiment of the sound effect processing method of the present invention;
FIG. 16 is a schematic structural diagram of an embodiment of a sound effect processing device according to the present invention;
fig. 17 is a schematic structural diagram of an embodiment of a terminal device provided in the present invention.
With the foregoing drawings in mind, certain embodiments of the disclosure have been shown and described in more detail below. These drawings and written description are not intended to limit the scope of the disclosed concepts in any way, but rather to illustrate the concepts of the disclosure to those skilled in the art by reference to specific embodiments.
Detailed Description
Reference will now be made in detail to the exemplary embodiments, examples of which are illustrated in the accompanying drawings. When the following description refers to the accompanying drawings, like numbers in different drawings represent the same or similar elements unless otherwise indicated. The implementations described in the exemplary embodiments below are not intended to represent all implementations consistent with the present disclosure. Rather, they are merely examples of devices consistent with certain aspects of the present disclosure, as detailed in the appended claims.
The terms "comprising" and "having," and any variations thereof, in the description and claims of this invention and the drawings described herein are intended to cover non-exclusive inclusions. For example, a process, method, system, article, or apparatus that comprises a list of steps or elements is not limited to only those steps or elements listed, but may alternatively include other steps or elements not listed, or inherent to such process, method, article, or apparatus.
First, the nouns and scenarios related to the present invention are introduced:
in general, listening to a stereo signal at the vertices of an equilateral triangle formed by a stereo speaker and a listener can obtain a high-quality stereo sound image. As shown in fig. 1, the first and second speakers are a left speaker and a right speaker, respectively, the fourth is a left sound field (around the left speaker), the fifth is a middle sound field, the sixth is a right sound field (around the right speaker), and the seventh is a surround sound field (including the surround areas of the left and right speakers).
When the positional relationship between the user and the two speakers is not an equilateral triangle, the quality of the middle position of the sound image is degraded. As shown in fig. 2, when the left and right speaker distances are enlarged (or the distance of the user from the connecting lines of the left and right speakers is reduced), the quality of the intermediate sound field may be degraded. As shown in fig. 3, when the left and right speaker distances decrease (or the distance between the user and the connecting lines of the left and right speakers increases), the surround sound field of the stereo sound decreases.
To address this problem, multiple channels may be employed, such as 5.1 or 7.1 channels, with a special intermediate channel added to the center speaker. The signal output to the center speaker is extracted from the stereo signal by signal processing.
In the related art, signals of left and right channels in a stereo signal may be added up and divided by 2, and the resultant signals may be output to a center speaker. However, the left and right channel signals originally include position information of the left and right sides, so that the sound localization quality is poor after simple processing; alternatively, a method of extracting an intermediate channel using Fast Fourier Transform (FFT) is used. In these methods, frequency analysis is performed by short-time fourier transform to grasp the frequency characteristics of the signal, and the intermediate channel signal is separated by filtering. However, these methods are very disadvantageous for real-time processing due to large amount of calculation and large resource consumption.
According to the method provided by the embodiment of the invention, the input sound signal is filtered to separate a first sound field signal, namely an intermediate sound field signal, and separate a second sound field signal, the first sound field signal is sent to the central loudspeaker, and the second sound field signal is sent to the corresponding loudspeaker, so that the problem that the quality of a sound image is reduced due to the change of the position of a user relative to the left loudspeaker and the right loudspeaker is solved.
The technical solution of the present invention will be described in detail below with specific examples. The following several specific embodiments may be combined with each other, and details of the same or similar concepts or processes may not be repeated in some embodiments.
In the embodiment of the present invention, a stereo signal is taken as an example for explanation, and each input sound signal may be a sound signal of a left channel or a sound signal of a right channel of the stereo signal.
As shown in fig. 4, the input sound signal of the left channel is divided into a middle sound field signal and other sound field signals except for the middle sound field, and the input sound signal of the right channel is divided into a middle sound field signal and other sound field signals except for the middle sound field.
Wherein, Lch is LL + LC; rch RR + RC.
Where Lch is the sound signal of the left channel, Rch is the sound signal of the right channel, LC and RC are the intermediate sound field signals, and LL and RR are the other sound field signals except the intermediate sound field.
The sound signals of the left and right sound channels of the two channels are further separated into two signals respectively, so that signals of four channels are obtained, and then each separated signal can be processed independently.
An example of an instrument distribution is shown in fig. 5. These instruments rely on the intensity difference between the left and right channels for localization.
1. For signals with the same amplitude and phase in the left and right channels, the mid sound field (position indicated by the fifth in fig. 1) is localized.
2. Signals of the same phase and different amplitudes in the left and right channels are localized to positions between the loudspeakers and the intermediate position.
3. The independent signals of each sound channel are positioned to the sound field position of each loudspeaker (such as the positions shown by the (r) and ((c)) in the figure 1); the independent signal refers to a signal independent from each other in each channel, that is, a signal without any relationship.
4. The uncorrelated signals within each channel are localized to the surround sound field positions (positions shown in fig. 1 c) of the loudspeakers; here, the uncorrelated signal refers to a signal having no linear relationship in each channel.
Wherein, the first type signal in 1 has strong correlation (the correlation coefficient is 1); 2, the correlation coefficient of the second type signal is between 0 and 1; the third and fourth signals of 3 and 4 are uncorrelated and have a correlation coefficient of 0.
On the basis of the above characteristics, the embodiments of the present invention separate signals by correlation between two stereo signals. Through signal processing, the first, third and fourth types of signals can be completely separated, the part with strong correlation characteristics in the second type of signals is classified into the first type of signals, and the part with weak correlation is classified into the third and fourth types of signals. The intermediate sound field signal is characterized by unclear sound source positions, i.e., strongly correlated sound signals, i.e., strongly correlated signals localized in the intermediate sound field. Objects that are weakly correlated, or independent, are separated into left and right channels for independent processing.
The following description will be given by taking only one input sound signal as an example:
FIG. 6 is a flowchart illustrating a sound processing method according to an embodiment of the present invention. As shown in fig. 6, the method provided by this embodiment includes:
Specifically, the input sound signal is divided into at least two signals, and the input sound signal can be copied to obtain at least two signals. The at least two signals obtained by copying are the same signals.
As shown in fig. 7, the left channel audio signal is taken as an example to be described, the input audio signal is denoted as x (n), and the first channel signal is filtered to obtain a first sound-field signal y (n), i.e., an intermediate sound-field signal.
As shown in fig. 8, the coefficient of the filter used in the filtering process is obtained by an adaptive algorithm based on the sound signal of the channel generating the first sound-field signal, and the sound signals of the other channels, i.e., the output signal y (n) of the filter and the input signal x (n) and the sound signal d (n) of the other channels have strong correlation.
And 603, delaying the second path of signal in the at least two paths of signals, and subtracting the first sound field signal from the delayed signal to obtain a second sound field signal.
As shown in fig. 7, the second path of signal is delayed to obtain delayed signal x (n) Z-nDelaying the processed signal x (n) Z-nThe first sound field signal y (n) is subtracted to obtain the second sound field signal, i.e. the signal other than the intermediate sound field signal. Wherein n is the number of delayed sampling points.
And step 604, sending the first sound field signal to a central loudspeaker, and sending the second sound field signal to a corresponding loudspeaker.
And sending the first sound field signal after the signal processing to a central loudspeaker, and sending the second sound field signal to a corresponding loudspeaker, for example, sending the second sound field signal after the sound signal processing of the left channel to the left loudspeaker, and sending the second sound field signal after the sound signal processing of the right channel to the right loudspeaker.
In the method of this embodiment, each input sound signal is divided into at least two signals; filtering a first path of signal in the at least two paths of signals to obtain a first sound field signal; the coefficient of a filter used for filtering is obtained by using a self-adaptive algorithm according to a first sound field signal and sound signals of at least two sound channels, so that the correlation coefficient of the sound signals of the at least two sound channels and the first sound field signal is larger than a preset threshold value, the second channel of signal in the at least two channels of signals is subjected to delay processing, and the first sound field signal is subtracted from the delayed signal to obtain a second sound field signal; the first sound field signal is sent to the central loudspeaker, the second sound field signal is sent to the corresponding loudspeaker, the first sound field signal and the second sound field signal are separated from the input sound signal, the first sound field signal is sent to the central loudspeaker, the second sound field signal is sent to the corresponding loudspeaker, the first sound field signal and the second sound field signal can be processed independently, the first sound field signal is a part with strong correlation in the sound signals of at least two sound channels, and the sound image quality of a middle sound field is improved.
On the basis of the foregoing embodiment, optionally, in order to separate signals with strong correlation in different sound channels to obtain a first sound field signal, that is, to make the separated first sound field signal and each input sound signal have strong correlation, as shown in fig. 8, when the at least two signals are three signals, after filtering a first signal of the at least two signals to obtain the first sound field signal, the method further includes:
delaying the sound signal of the second channel and subtracting the first sound field signal to obtain an error signal; the first sound field signal is obtained after the sound signal of the first sound channel is filtered;
and adjusting the coefficient of the filter adopted by the filtering processing by adopting a self-adaptive algorithm according to a third signal in the three signals and the error signal.
Specifically, as shown in fig. 8, the sound signal of the left channel is taken as an example to describe, the input sound signal is denoted as x (n), three paths of signals are obtained by copying, the first path of signal is filtered to obtain a first sound field signal y (n), which is an intermediate sound field signal LC, and the second path of signal is delayed to obtain a delayed signal x (n) Z-nDelaying the processed signal x (n) Z-nThe first sound field signal y (n) is subtracted to obtain a second sound field signal, i.e. signal LL, except for the intermediate sound field signal.
Then, the sound signal of the right channel is delayed to obtain d (n) Z-nSubtracting the first sound field signal to obtain an error signal e (n), e (n) ═ d (n) Z-n-y(n)。
And adjusting the coefficients of the filter by adopting a self-adaptive algorithm for the third path of signal and the error signal so as to minimize the error signal. Wherein, when the output signal y (n) of the FIR filter is strongly correlated with the input signal x (n) and the sound signal d (n) of the second channel, the error signal e (n) is minimized. Therefore, the first sound field signal obtained by the filter processing is a portion of the input sound signal having a strong correlation, and can be transmitted to the center speaker as an intermediate sound field signal.
FIG. 9 is a schematic diagram of an implementation of an FIR filter, where TAP is the number of TAPs of the filter and the coefficient of the filter is W0、W1、W2、…、WTAP-2、WTAP-1。
y(n)＝W0x(n)+W1x(n)Z-1+W2x(n)Z-2+…+WTAP-2x(n)Z-(TAP-2)+WTAP-1x(n))Z-(TAP-1)
Alternatively, the coefficients of the filter are updated every time a sample is processed.
Further, the number of sampling points n of the delay process may be equal to half of the number of taps of the filter.
The adaptive algorithm may adopt a Least Mean Square (LMS) algorithm, a Normalized Least Mean Square (NLMS) algorithm, a Recursive Least Square (RLS) algorithm, and the like.
In the figure, the filter used in the filtering process is a (Finite Impulse Response, abbreviated as FIR) filter, and in other embodiments of the present invention, the filter may also be implemented by an (Infinite Impulse Response, abbreviated as IIR) filter.
The processing procedure of the sound signal of the right channel in fig. 10 is similar to the processing procedure of the sound signal of the left channel, and is not repeated here. In FIG. 10, y (n) is the intermediate sound field signal RC, and the delayed signal x (n) Z-nThe first sound field signal y (n) is subtracted to obtain the second sound field signal, i.e., the signal RR except for the intermediate sound field signal.
As shown in fig. 11, the processing procedure of two signals for the left channel and the right channel may specifically refer to the foregoing embodiment, and is not described herein again.
In the embodiment of the invention, the separation of the intermediate sound field signals is realized through the filter, the coefficients of the filter are adjusted through the self-adaptive algorithm, the stereo signals are separated by using less computation and resources, and an independent intermediate sound channel can be added and an independent central loudspeaker can be added through the signal separation operation.
On the basis of the above embodiment, optionally, as shown in fig. 12, the sound signals include a sound signal of a first channel and a sound signal of a second channel; transmitting the first sound field signal to a center speaker in step 604, comprising:
carrying out weighting processing on a first sound field signal obtained after processing the sound signal of the first sound channel and a first sound field signal obtained after processing the sound signal of the second sound channel;
the weighted signal is sent to the center speaker.
Specifically, if the input sound signal is a stereo signal, the sound signal includes sound signals of two channels, and before the first sound field signal is sent to the speakers, a weighting process may be performed on the first sound field signal obtained by processing the sound signals of two channels, for example, as shown in fig. 12, the two first sound field signals are added, then multiplied by a coefficient of 0.5, and sent to the center speaker after the weighting process.
Fig. 13 shows a process of extracting the intermediate sound field signal in fig. 12.
In this embodiment, in order to extract the intermediate sound field signal CC having strong correlation from the left and right channels, the output signals LC, RC of the FIR filters corresponding to the left and right channels are subjected to an addition operation in which the signal amplitude is halved, i.e., multiplied by 0.5. Such processing enhances signals with strong correlation.
Further, the first sound field signal after the sound signal filtering processing of any one of the first sound channel and the second sound channel is processed by utilizing a signal enhancement processing algorithm or an HRTF correction algorithm;
and sending the processed signals to a loudspeaker corresponding to any sound channel.
Specifically, the signal enhancement processing algorithm or a Head Related Transfer Function (HRTF) correction algorithm may be used to perform signal enhancement processing on the first sound field signal (i.e., the intermediate sound field signal), so as to improve the quality of the intermediate sound field.
As shown in fig. 14, the first sound field signal LC obtained by processing the sound signal of the first channel may be subjected to signal enhancement processing, and sent to a speaker corresponding to the first channel (a speaker corresponding to the left channel in the figure, that is, a left speaker); the first sound field signal RC obtained by processing the sound signal of the second channel is subjected to signal enhancement processing, and is sent to a speaker corresponding to the second channel (a speaker corresponding to the right channel in the drawing, that is, a right speaker).
In this embodiment, the intermediate sound field signal obtained by separating the sound signal may be separately subjected to other signal processing, such as enhancement processing, which is better than the original process of mixing the left and right sound channels together, and the sound field quality may be improved.
On the basis of the foregoing embodiment, further, sending the second acoustic field signal to a corresponding speaker includes:
performing signal processing of a stereo sound image expansion effect or a surround effect on a second sound field signal obtained by processing a sound signal of any one of a first sound channel and a second sound channel to obtain a processed second sound field signal;
and sending the processed second sound field signal to a loudspeaker corresponding to any sound channel.
Specifically, independent signal processing may be performed on other sound field signals in addition to the intermediate sound field signal. Thus, even if the space between the left and right stereo speakers is enlarged, the surround sound effect can be improved without affecting the central sound field.
As shown in fig. 15, the sound signal of the left channel or the sound signal of the right channel is processed and transmitted to the corresponding speaker.
In this embodiment, the sound signals may be separated to obtain other sound field signals except the intermediate sound field signal, and the other signal processing may be performed separately, for example, signal processing of a stereo sound image extension effect or a surround effect is better than the original processing of mixing left and right channels together, so that the sound field quality may be improved.
Fig. 16 is a structural diagram of an audio processing device according to an embodiment of the present invention, and as shown in fig. 16, the audio processing device of the embodiment includes:
the preprocessing module 161 is configured to divide each input sound signal into at least two signals; the sound signals comprise sound signals of at least two sound channels;
the processing module 162 is configured to perform filtering processing on a first path of signal in the at least two paths of signals to obtain a first sound field signal; the coefficient of the filter adopted by the filtering processing is obtained by utilizing a self-adaptive algorithm according to the first sound field signal and the sound signals of other sound channels;
the processing module 162 is further configured to perform delay processing on a second signal of the at least two signals, and subtract the first sound field signal from the delayed signal to obtain a second sound field signal;
a sending module 163, configured to send the first sound field signal to a central speaker, and send the second sound field signal to a corresponding speaker.
Optionally, the sound signals of the at least two channels are a sound signal of a first channel and a sound signal of a second channel; the sending module 163 is specifically configured to:
weighting a first sound field signal obtained by processing the sound signal of the first sound channel and a first sound field signal obtained by processing the sound signal of the second sound channel;
and sending the weighted signals to the central loudspeaker.
Optionally, when the at least two signals are three signals, the processing module 162 is further specifically configured to:
delaying the sound signal of the second channel and subtracting the first sound field signal to obtain an error signal; the first sound field signal is obtained after the sound signal of the first sound channel is filtered;
and adjusting the coefficient of a filter adopted by filtering processing by adopting a self-adaptive algorithm according to a third signal in the three signals and the error signal so as to minimize the error signal.
Optionally, the processing module 162 is further configured to:
processing a first sound field signal obtained by filtering the sound signal of any one of the first channel and the second channel by using a signal enhancement processing algorithm or an HRTF correction algorithm;
and sending the processed signal to a loudspeaker corresponding to any sound channel.
Optionally, the sending module 163 is specifically configured to:
performing signal processing of a stereo sound image expansion effect or a surround effect on a second sound field signal obtained by processing the sound signal of any one of the first sound channel and the second sound channel to obtain a processed second sound field signal;
and sending the processed second sound field signal to a loudspeaker corresponding to any sound channel.
Optionally, the processing module 162 is specifically configured to:
and performing FIR filtering or IIR filtering processing on the first path of signals to obtain the first sound field signals.
Optionally, the number of sampling points of the delay processing is equal to half of the number of taps of the filter used in the filtering processing.
The apparatus of this embodiment may be configured to implement the technical solutions of the above method embodiments, and the implementation principles and technical effects are similar, which are not described herein again.
Fig. 17 is a structural diagram of an embodiment of a terminal device provided in the present invention, and as shown in fig. 17, the terminal device includes:
a processor 171, and a memory 172 for storing executable instructions for the processor 171.
The processor 171 is configured to execute the corresponding method in the foregoing method embodiment by executing the executable instruction, and the specific implementation process thereof may refer to the foregoing method embodiment, which is not described herein again.
The embodiment of the present invention further provides a computer-readable storage medium, where a computer program is stored, and when the computer program is executed by a processor, the method in the foregoing method embodiment is implemented.
Other embodiments of the disclosure will be apparent to those skilled in the art from consideration of the specification and practice of the disclosure disclosed herein. This application is intended to cover any variations, uses, or adaptations of the disclosure following, in general, the principles of the disclosure and including such departures from the present disclosure as come within known or customary practice within the art to which the disclosure pertains. It is intended that the specification and examples be considered as exemplary only, with a true scope and spirit of the disclosure being indicated by the following claims.
It will be understood that the present disclosure is not limited to the precise arrangements described above and shown in the drawings and that various modifications and changes may be made without departing from the scope thereof. The scope of the present disclosure is limited only by the appended claims.
Claims (9)
1. A sound effect processing method is characterized by comprising the following steps:
dividing each path of input sound signals into at least two paths of signals; the sound signals comprise sound signals of at least two sound channels;
filtering a first path of signal in the at least two paths of signals to obtain a first sound field signal; the coefficients of the filter adopted in the filtering process are obtained by using an adaptive algorithm according to the first sound field signal and the sound signals of the at least two sound channels, so that the correlation coefficients of the sound signals of the at least two sound channels and the first sound field signal are greater than a preset threshold;
delaying a second path of signal in the at least two paths of signals, and subtracting the first sound field signal from the delayed signal to obtain a second sound field signal;
and sending the first sound field signal to a central loudspeaker, and sending the second sound field signal to a corresponding loudspeaker.
2. The method according to claim 1, wherein the sound signals of the at least two channels are a sound signal of a first channel and a sound signal of a second channel; the sending the first soundfield signal to a center speaker, comprising:
weighting a first sound field signal obtained by processing the sound signal of the first sound channel and a first sound field signal obtained by processing the sound signal of the second sound channel;
and sending the weighted signals to the central loudspeaker.
3. The method according to claim 2, wherein when the at least two signals are three-way signals, after filtering a first signal of the at least two signals to obtain a first sound field signal, the method further comprises:
delaying the sound signal of the second channel and subtracting the first sound field signal to obtain an error signal; the first sound field signal is obtained after the sound signal of the first sound channel is filtered;
and adjusting the coefficient of a filter adopted by filtering processing by adopting a self-adaptive algorithm according to a third signal in the three signals and the error signal so as to minimize the error signal.
4. The method of claim 2 or 3, wherein sending the second acoustic field signal to a corresponding speaker comprises:
performing signal processing of a stereo sound image expansion effect or a surround effect on a second sound field signal obtained by processing the sound signal of any one of the first sound channel and the second sound channel to obtain a processed second sound field signal;
and sending the processed second sound field signal to a loudspeaker corresponding to any sound channel.
5. The method according to any one of claims 1-3, wherein the filtering the first signal of the at least two signals to obtain the first sound field signal comprises:
and performing FIR filtering or IIR filtering processing on the first path of signals to obtain the first sound field signals.
6. The method according to any one of claims 1 to 3,
the number of sampling points of the delay processing is equal to half of the number of taps of the filter adopted by the filtering processing.
7. An audio processing apparatus, comprising:
the preprocessing module is used for dividing each path of input sound signals into at least two paths of signals; the sound signals comprise sound signals of at least two sound channels;
the processing module is used for filtering a first path of signal in the at least two paths of signals to obtain a first sound field signal; the coefficients of the filter adopted in the filtering process are obtained by using an adaptive algorithm according to the first sound field signal and the sound signals of the at least two sound channels, so that the correlation coefficients of the sound signals of the at least two sound channels and the first sound field signal are greater than a preset threshold;
the processing module is further configured to perform delay processing on a second path of signals in the at least two paths of signals, and subtract the first sound field signal from the delayed signals to obtain a second sound field signal;
and the sending module is used for sending the first sound field signal to a central loudspeaker and sending the second sound field signal to a corresponding loudspeaker.
8. The apparatus of claim 7, wherein the sound signals of the at least two channels are a sound signal of a first channel and a sound signal of a second channel; the sending module is specifically configured to:
weighting a first sound field signal obtained by processing the sound signal of the first sound channel and a first sound field signal obtained by processing the sound signal of the second sound channel;
and sending the weighted signals to the central loudspeaker.
9. The apparatus of claim 8, wherein when the at least two signals are three signals, the processing module is further specifically configured to:
delaying the sound signal of the second channel and subtracting the first sound field signal to obtain an error signal; the first sound field signal is obtained after the sound signal of the first sound channel is filtered;
and adjusting the coefficient of a filter adopted by filtering processing by adopting a self-adaptive algorithm according to a third signal in the three signals and the error signal so as to minimize the error signal.
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