CN108810737B - Signal processing method and device and virtual surround sound playing equipment - Google Patents
Signal processing method and device and virtual surround sound playing equipment Download PDFInfo
- Publication number
- CN108810737B CN108810737B CN201810282945.5A CN201810282945A CN108810737B CN 108810737 B CN108810737 B CN 108810737B CN 201810282945 A CN201810282945 A CN 201810282945A CN 108810737 B CN108810737 B CN 108810737B
- Authority
- CN
- China
- Prior art keywords
- channel
- signal
- sampling point
- input
- sound
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Active
Links
- 238000003672 processing method Methods 0.000 title description 6
- 238000005070 sampling Methods 0.000 claims abstract description 138
- 238000000034 method Methods 0.000 claims abstract description 41
- 238000009432 framing Methods 0.000 claims abstract description 19
- 230000000694 effects Effects 0.000 abstract description 14
- 238000010586 diagram Methods 0.000 description 20
- 210000005069 ears Anatomy 0.000 description 10
- 230000008569 process Effects 0.000 description 10
- 230000005540 biological transmission Effects 0.000 description 7
- 238000004590 computer program Methods 0.000 description 7
- 230000006870 function Effects 0.000 description 7
- 238000005516 engineering process Methods 0.000 description 5
- 230000009471 action Effects 0.000 description 4
- 210000004556 brain Anatomy 0.000 description 4
- 230000003111 delayed effect Effects 0.000 description 3
- 210000003128 head Anatomy 0.000 description 3
- 230000009977 dual effect Effects 0.000 description 2
- 238000004519 manufacturing process Methods 0.000 description 2
- 238000012986 modification Methods 0.000 description 2
- 230000004048 modification Effects 0.000 description 2
- 230000004075 alteration Effects 0.000 description 1
- 230000000903 blocking effect Effects 0.000 description 1
- 230000008859 change Effects 0.000 description 1
- 230000004807 localization Effects 0.000 description 1
- 230000003287 optical effect Effects 0.000 description 1
- 230000000750 progressive effect Effects 0.000 description 1
- 230000005236 sound signal Effects 0.000 description 1
- 230000005654 stationary process Effects 0.000 description 1
- 210000003454 tympanic membrane Anatomy 0.000 description 1
Images
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R3/00—Circuits for transducers, loudspeakers or microphones
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
- H04R2430/00—Signal processing covered by H04R, not provided for in its groups
Landscapes
- Physics & Mathematics (AREA)
- Engineering & Computer Science (AREA)
- Acoustics & Sound (AREA)
- Signal Processing (AREA)
- Stereophonic System (AREA)
Abstract
The embodiment of the invention provides a method and a device for processing signals and virtual surround sound playing equipment, wherein the method is applied to the virtual surround sound playing equipment and comprises the following steps: receiving an input signal; framing the signal to obtain a signal to be processed, wherein the signal to be processed comprises a first sound channel sampling point signal and a second sound channel sampling point signal; generating a first channel to-be-input signal based on the first channel sampling point signal; carrying out phase inversion operation on the second channel sampling point signal, and generating a second channel signal to be input based on the second channel sampling point signal after phase inversion; inputting the first channel signal to be input to a first channel loudspeaker; and inputting the second channel to-be-input signal to a second channel loudspeaker to obtain a virtual surround sound effect.
Description
Technical Field
The present invention relates to the field of signal processing technologies, and in particular, to a signal processing method, a signal processing apparatus, and a virtual surround sound playing device.
Background
Virtual Surround (Virtual Surround) technology is a sound technology that can process and play back multichannel signals in two parallel speakers and make people feel the effect of Surround sound, and is widely used in the production of televisions. The application of the virtual surround sound technology in the television is mainly to enlarge the sound field of the television and make the sound of the television sound with a sense of containment.
Currently, the virtual surround sound technology is mainly implemented by an HRTF (head-related transfer function) algorithm. However, the effect of the algorithm has a large relationship with the position of the user in front of the television, and when the position of the user changes, the obtained sound effect also changes greatly. Fig. 1 is a schematic diagram of positioning a point sound source by human ears. In general, the localization of a sound source by the human ear is mainly based on three aspects of information. One is the time difference (also called phase difference) of arrival of the sound source at the two ears; secondly, the sound intensity difference of the sound source reaching the two ears; the third is the change in the signals arriving at the eardrums of the two ears by different transmission paths due to the structure of the head and the ears. The human brain can deduce the specific position of the sound source by integrating the information of the three aspects. However, when a person faces a sound source, it is difficult to infer the specific location of the sound source from the three information.
Fig. 2 is a schematic diagram of positioning a face sound source by human ears. When a person stands in front of a wall with speakers hanging on one side and the speakers simultaneously emit the same sound, signals reaching two ears of the person include complex delay, phase and other signals filtered by the head and the ears, so that the brain of the person is difficult to locate a specific direction of a certain sound source, can only feel the approximate position of a sound field with a large range, and cannot effectively experience an obvious virtual surround sound effect.
Disclosure of Invention
In view of the above problems, embodiments of the present invention are proposed to provide a method of signal processing, an apparatus for signal processing and a corresponding virtual surround sound playing device that overcome or at least partially solve the above problems.
In order to solve the above problem, an embodiment of the present invention discloses a signal processing method, which is applied to a virtual surround sound playing device, and the method includes:
receiving an input signal;
framing the signal to obtain a signal to be processed, wherein the signal to be processed comprises a first sound channel sampling point signal and a second sound channel sampling point signal;
generating a first channel to-be-input signal based on the first channel sampling point signal; and the number of the first and second groups,
carrying out phase inversion operation on the second channel sampling point signal, and generating a second channel signal to be input based on the second channel sampling point signal after phase inversion;
inputting the first channel signal to be input to a first channel loudspeaker;
and inputting the second channel to-be-input signal to a second channel loudspeaker.
Optionally, the step of framing the signal to obtain a signal to be processed includes:
determining the frame length of the framed signal;
and dividing the signal into a plurality of frames of signals to be processed according to the frame length.
Optionally, the first channel signal to be input includes a first channel weighting signal and multiple first channel band pass signals, and the step of generating the first channel signal to be input based on the first channel sampling point signal includes:
determining a plurality of frequency bands of the first channel sampling point signal;
transmitting the first sound channel sampling point signal to a band-pass filter of a corresponding frequency band;
respectively carrying out delay and weighting processing on the first sound channel sampling point signals to obtain a plurality of paths of first sound channel band-pass signals; and the number of the first and second groups,
and carrying out weighting processing on the first channel sample point signal to obtain a first channel weighted signal.
Optionally, the step of respectively delaying and weighting the first channel sample point signals to obtain multiple first channel band pass signals includes:
determining a delay time and a weighting coefficient of the first channel band-pass signal;
adding a plurality of leading zeros in the first channel sample point signal according to the delay time;
and multiplying the first channel sampling point signal after delay processing by the weighting coefficient to keep the total volume of the first channel sampling point signal unchanged.
Optionally, the multiple frequency bands include a high frequency, an intermediate frequency, and a low frequency, where a delay time of the high frequency first channel sampling signal is less than a delay time of the intermediate frequency first channel sampling signal, and a delay time of the intermediate frequency first channel sampling signal is less than a delay time of the low frequency first channel sampling signal.
Optionally, the step of performing an inversion operation on the second channel sample signal includes:
and inverting the second channel sampling point signal through a preset inverter to obtain an inverted second channel sampling point signal.
Optionally, the second channel to-be-input signal includes a second channel weighted signal and multiple second channel bandpass signals, and the step of generating the second channel to-be-input signal based on the inverted second channel sampling point signal includes:
determining a plurality of frequency bands of the second channel sampling point signal after phase inversion;
transmitting the second channel sampling point signal after phase inversion to a band-pass filter of a corresponding frequency band;
respectively carrying out delay and weighting processing on the second channel sampling point signals after phase inversion to obtain a plurality of paths of second channel band-pass signals; and the number of the first and second groups,
and carrying out weighting processing on the second channel sampling point signal after phase inversion to obtain a second channel weighted signal.
Optionally, if the first channel is a left channel, the second channel is a right channel; if the first channel is a right channel, the second channel is a left channel.
In order to solve the above problem, an embodiment of the present invention discloses a signal processing apparatus, which is applied to a virtual surround sound playing device, and includes:
the receiving module is used for receiving an input signal;
the framing module is used for framing the signal to obtain a signal to be processed, wherein the signal to be processed comprises a first sound channel sampling point signal and a second sound channel sampling point signal;
a first generating module, configured to generate a first channel to-be-input signal based on the first channel sampling point signal; and the number of the first and second groups,
the second generation module is used for carrying out phase inversion operation on the second channel sampling point signal and generating a second channel signal to be input based on the second channel sampling point signal after phase inversion;
the first input module is used for inputting the first channel to-be-input signal to a first channel loudspeaker;
and the second input module is used for inputting the signal to be input in the second channel to the loudspeaker in the second channel.
Optionally, the framing module includes:
the frame length determining submodule is used for determining the frame length of the framed signal;
and the signal framing submodule is used for dividing the signal into a plurality of frames of signals to be processed according to the frame length.
Optionally, the first channel signal to be input includes a first channel weighting signal and multiple first channel band pass signals, and the first generating module includes:
a first determining sub-module, configured to determine a plurality of frequency bands of the first channel sampling point signal;
the first transmission submodule is used for transmitting the first sound channel sampling point signal to a band-pass filter of a corresponding frequency band;
the first delay weighting processing sub-module is used for respectively carrying out delay and weighting processing on the first sound channel sampling point signals so as to obtain a plurality of paths of first sound channel band-pass signals; and the number of the first and second groups,
and the first weighting submodule is used for weighting the first channel sample point signal to obtain a first channel weighted signal.
Optionally, the first delay weighting processing sub-module includes:
a first determining unit for determining a delay time and a weighting coefficient of the first channel band pass signal;
a first delay unit, configured to add a plurality of leading zeros in the first channel sample signal according to the delay time;
and the first weighting unit is used for multiplying the first channel sampling point signal after delay processing by the weighting coefficient so as to keep the total volume of the first channel sampling point signal unchanged.
Optionally, the multiple frequency bands include a high frequency, an intermediate frequency, and a low frequency, where a delay time of the high frequency first channel sampling signal is less than a delay time of the intermediate frequency first channel sampling signal, and a delay time of the intermediate frequency first channel sampling signal is less than a delay time of the low frequency first channel sampling signal.
Optionally, the second generating module includes:
and the negation submodule is used for negating the second channel sampling point signal through a preset phase inverter to obtain a second channel sampling point signal after phase inversion.
Optionally, the second channel to-be-input signal includes a second channel weighted signal and multiple second channel band-pass signals, and the second generating module includes:
the second determining submodule is used for determining a plurality of frequency bands of the second channel sampling point signals after phase inversion;
the second transmission submodule is used for transmitting the second channel sampling point signal after phase inversion to a band-pass filter of a corresponding frequency band;
the second delay weighting processing submodule is used for respectively carrying out delay and weighting processing on the second channel sampling point signals after phase inversion so as to obtain a plurality of paths of second channel on-band signals; and the number of the first and second groups,
and the second weighting submodule is used for weighting the second channel sampling point signal after phase inversion so as to obtain a second channel weighting signal.
Optionally, if the first channel is a left channel, the second channel is a right channel; if the first channel is a right channel, the second channel is a left channel.
In order to solve the above problem, an embodiment of the present invention discloses a virtual surround sound playing device, which includes a first channel speaker, a second channel speaker, and the above signal processing apparatus.
Compared with the background art, the embodiment of the invention has the following advantages:
in the embodiment of the invention, after an input signal is received, firstly, the signal is framed to obtain a first sound channel sampling point signal and a second sound channel sampling point signal to be processed, then, a first sound channel signal to be input is generated based on the first sound channel sampling point signal, and a second sound channel signal to be input is generated based on the second sound channel sampling point signal after phase inversion; and respectively inputting the first channel to-be-input signal and the second channel to-be-input signal into a first channel loudspeaker and a second channel loudspeaker for playing to obtain a virtual surround sound effect. In the embodiment, the signal input into one of the sound channel speakers is directly processed, and the signal input into the other sound channel speaker is processed after the phase inversion operation is performed, so that a sound field with multiple sound intensity differences and phase differences can be generated in front of the multiple speakers, the left ear and the right ear of a person in the sound field can hear sounds with different complex phase differences and amplitude differences as far as possible, the effect of virtual surround sound is obtained, and the telepresence of a user is greatly enhanced.
Drawings
FIG. 1 is a diagram of a human ear positioning a point source of sound in the prior art;
FIG. 2 is a diagram of a prior art human ear locating a face sound source;
FIG. 3 is a flow chart illustrating steps of a method of signal processing according to an embodiment of the present invention;
FIG. 4 is a schematic diagram of a sound field implementation principle of a method of signal processing according to an embodiment of the present invention;
FIG. 5 is a flow chart illustrating steps of another method of signal processing according to one embodiment of the present invention;
FIG. 6 is a schematic diagram of a method of signal processing according to an embodiment of the present invention;
FIG. 7 is a block diagram of a schematic configuration of a signal processing apparatus according to an embodiment of the present invention;
fig. 8 is a schematic block diagram of a virtual surround sound playing apparatus according to an embodiment of the present invention.
Detailed Description
In order to make the aforementioned objects, features and advantages of the present invention comprehensible, embodiments accompanied with figures are described in further detail below.
Referring to fig. 3, a schematic flow chart illustrating steps of a method for signal processing according to an embodiment of the present invention is shown, which may specifically include the following steps:
it should be noted that the method can be applied to various audio playing devices, such as a virtual surround sound playing device. The device may be a stand-alone audio player, such as various types of audio equipment; other devices integrating an audio playing function, such as a television, may also be used, and this embodiment is not limited in this respect.
In general, the audio playback apparatus may include at least two speakers, i.e., a first channel speaker and a second channel speaker. If the first sound channel is a left sound channel, the second sound channel is a right sound channel; if the first channel is a right channel, the second channel is a left channel.
In the embodiment of the present invention, the input signal may be a digital voice signal, and the digital voice signal may be converted into an audio signal after being encoded, decoded, and processed in a series of other processing manners, and output through a speaker, etc., so that a user can hear a corresponding audio.
generally, the characteristics of a speech signal as a whole and the parameters characterizing its essential features are time-varying, so it is a non-stationary process. But in a short time range (generally considered to be 10-30ms (milliseconds)), the characteristics can be basically kept unchanged, i.e. relatively stable, and thus the method can be regarded as a quasi-steady-state process, i.e. the speech signal has short-term stationarity.
Therefore, the speech signal can be divided into segments to analyze its characteristic parameters, wherein each segment is called a "frame", and the length of the frame can be typically 10-30 ms. Thus, for the whole speech signal, the analyzed characteristic parameter time sequence is composed of the characteristic parameters of each frame.
In the embodiment of the present invention, after receiving an input signal, the signal may be subjected to framing processing to obtain a frame-by-frame signal to be processed. The signal to be processed may include a first channel sample signal and a second channel sample signal. The first channel sample signal is a signal that is processed and transmitted to the first channel speaker, and the second channel sample signal is a signal that is processed and transmitted to the second channel speaker.
fig. 4 is a schematic diagram illustrating a sound field implementation principle of a virtual surround sound playing method according to an embodiment of the present invention. When a user faces the front of a sound source formed by a plurality of loudspeakers and the loudspeakers simultaneously emit the same sound, a sound field with multiple sound intensity differences and phase differences can be generated in front of the loudspeakers through the action of signal processing and wall surface reflection, so that the left ear and the right ear of a person in the sound field can hear the sound with the complex phase differences and amplitude differences which are different as much as possible, and the effect of virtual surround sound is achieved.
In a specific implementation, the signal to be transmitted to one of the speakers in the input signal may be directly processed, and the signal to be transmitted to the other speaker may be first subjected to the inverse phase operation and then subjected to the same processing as the signal to be transmitted to the other speaker. For example, the signal to be fed to the left channel speaker is directly processed, while the signal to be fed to the right channel speaker is first inverted and then processed. After processing, the signals can be sent to left and right speakers respectively, and then reach the left and right ears of a person through different transmission paths. At this moment, because the signals in the left and right sound channel loudspeakers contain a large number of delay signals with different intensities, and in addition, the sound waves are reflected on the wall surface for a plurality of times, the phase and sound intensity of different position points in the sound field in front of the loudspeakers are obviously changed, so that the sound heard by the left and right ears in the sound field is greatly different, and the difference of a plurality of phases and sound intensities is contained, so that a surface sound source can be simulated, and an obvious virtual surround sound effect is obtained.
Therefore, in the embodiment of the present invention, the first channel sample signal may be directly processed after being received.
In a specific implementation, the processing of the first channel sample signals may be performed in two ways. And part of the first channel sample point signals are directly weighted to generate first channel weighted signals, so that the sound quality of the sound is ensured and the original characteristics of the sound are maintained. On the other hand, all the first channel sampling point signals are input into corresponding band-pass filters, and then processing such as delaying and weighting is carried out to obtain multi-channel first channel band-pass signals, so that the complexity of the signals is enhanced.
A band-pass filter is a device that allows waves in a particular frequency band to pass through while blocking the transmission of waves in other frequency bands. For example, for a high frequency band pass filter, only high frequency signals can pass through, and intermediate and low frequency signals cannot pass through; for a low-frequency band-pass filter, only low-frequency signals can be allowed to pass, and high-frequency and intermediate-frequency signals cannot pass.
in the embodiment of the present invention, the processing for the second channel sampling signals is basically the same as the processing for the first channel sampling signals, and the only difference is that all the second channel sampling signals need to be inverted before being processed.
That is, first, the phase inversion operation is performed on all the second channel sampling signals, and then the same processing steps as those of the first channel sampling signals are performed on the second channel sampling signals after the phase inversion, so as to obtain a second channel weighted signal and a plurality of second channel band-pass signals.
in the embodiment of the present invention, the first channel signal to be input includes the first channel weighting signal and the plurality of first channel band pass signals generated in step 303.
After the first channel weighted signal and the plurality of first channel band pass signals are obtained, the first channel weighted signal and the plurality of first channel band pass signals can be input to a first channel loudspeaker and played through the first channel loudspeaker.
And step 306, inputting the second channel signal to be input to a second channel loudspeaker.
In the embodiment of the present invention, the second channel signal to be input includes the second channel weighting signal and the plurality of second channel bandpass signals generated in step 304.
After the second channel weighting signal and the multi-channel second channel band-pass signal are obtained, the second channel weighting signal and the multi-channel second channel band-pass signal can be input to a second channel loudspeaker and played through the second channel loudspeaker, so that an obvious virtual surround sound effect can be obtained under the cooperation of the first channel loudspeaker and the second channel loudspeaker.
In the embodiment of the invention, after receiving an input signal, firstly framing the signal to obtain a first channel sampling point signal and a second channel sampling point signal to be processed, then generating a first channel signal to be input based on the first channel sampling point signal, and generating a second channel signal to be input based on the second channel sampling point signal after phase inversion; and respectively inputting the first channel to-be-input signal and the second channel to-be-input signal into a first channel loudspeaker and a second channel loudspeaker for playing to obtain a virtual surround sound effect. In the embodiment, the signal input into one of the sound channel speakers is directly processed, and the signal input into the other sound channel speaker is processed after the phase inversion operation is performed, so that a sound field with multiple sound intensity differences and phase differences can be generated in front of the multiple speakers, the left ear and the right ear of a person in the sound field can hear sounds with different complex phase differences and amplitude differences as far as possible, the effect of virtual surround sound is obtained, and the telepresence of a user is greatly enhanced.
Referring to fig. 5, a schematic flow chart illustrating steps of another signal processing method according to an embodiment of the present invention is shown, which may specifically include the following steps:
it should be noted that the method can be applied to various audio playing devices, such as a virtual surround sound playing device. The device may be a stand-alone audio player, such as various types of audio equipment; other devices integrating an audio playing function, such as a television, may also be used, and this embodiment is not limited in this respect.
For ease of understanding, the following description of the present embodiment takes a television as an example of the audio playing device.
In general, a television set may include two speakers, a first channel speaker and a second channel speaker. If the first sound channel is a left sound channel, the second sound channel is a right sound channel; if the first channel is a right channel, the second channel is a left channel.
For convenience of understanding, the following description will be given by taking the first channel speaker as a left channel speaker and the second channel speaker as a right channel speaker as an example.
in general, the characteristics of a speech signal can remain substantially unchanged, i.e., relatively stable, over a short time frame, and thus can be considered as a quasi-stationary process, i.e., the speech signal has short-term stationarity.
Fig. 6 is a schematic diagram illustrating an implementation process of a method for signal processing according to an embodiment of the present invention. In fig. 6, after receiving an input signal, the signal may be subjected to framing processing to obtain a frame-by-frame signal to be processed.
In a specific implementation, the frame length of the framed signal may be determined first, and then the signal may be divided into multiple frames of signals to be processed according to the frame length. Typically, the frame length of each frame may be between 10-30 ms.
In an embodiment of the present invention, the signal to be processed may include a first channel sample signal and a second channel sample signal, that is, a left channel sample signal and a right channel sample signal.
in the embodiment of the present invention, the first channel sample signal is a left channel sample signal. The processing of the left channel sampling point signals can be divided into two aspects, wherein on one hand, all the left channel sampling point signals can be input into corresponding band pass filters, and then the processing such as delaying, weighting and the like is carried out to obtain a plurality of paths of left channel band pass signals, so that the complexity of the signals is enhanced.
In a specific implementation, a plurality of frequency bands of the left channel sampling point signal may be determined respectively, so as to set a band pass filter of the corresponding frequency band.
For example, as shown in fig. 6, the frequency bands of the left channel sampling point signal may be respectively set to a high frequency, an intermediate frequency, and a low frequency, and then three band pass filters of the high frequency band pass filter, the intermediate frequency band pass filter, and the low frequency band pass filter may be provided.
Of course, according to actual needs, those skilled in the art may also select to set a greater number of bandpass filters for each specific frequency band, and the number of the bandpass filters set in this embodiment is not limited.
it should be noted that, when the left channel sampling point signal of each frequency band passes through the band pass filter, the left channel sampling point signal can only pass through the band pass filter of the corresponding frequency band, but cannot pass through the band pass filters of other frequency bands. That is to say, through the processing of the band-pass filter, the sampling point signals of different frequency bands can be distinguished, which is convenient for the subsequent further processing.
For example, a high-frequency sampling signal can only pass through a high-frequency band-pass filter, but cannot pass through an intermediate-frequency band-pass filter and a low-frequency band-pass filter; for low-frequency sampling signals, only the low-frequency band-pass filter can pass through, but the intermediate-frequency band-pass filter and the high-frequency band-pass filter cannot pass through.
Step 505, respectively performing delay and weighting processing on the first channel sampling point signals to obtain multiple paths of first channel band-pass signals;
as shown in fig. 6, the processing of the left channel sample signals of the respective frequency bands may include delay processing and weighting processing. The band-pass signals of different frequency bands are selected by each band-pass signal, so that the complexity of the signals can be increased.
In the embodiment of the present invention, when processing the sampling point signals of each frequency band, first, a delay time and a weighting coefficient of the first channel bandpass signal are respectively determined, then, according to the delay time, a plurality of leading zeros are added to the first channel sampling point signals of each frequency band, and the first channel sampling point signals of each frequency band after the delay processing are multiplied by the weighting coefficient, so that the total volume of the first channel sampling point signals is kept unchanged. Leading zeros are a display format, and in the embodiment of the present invention, if the delay time of the first channel sample point signal of a certain frequency band is longer, more leading zeros may be added in front of the signal, and if the delay time is shorter, less leading zeros may be added. The number of leading zeros added in front of the first channel sample signal may be determined according to the actually determined delay time, which is not limited in this embodiment.
In a specific implementation, in the delay processing, the general trend is that the delay time of the low-frequency signal is longer, the delay time of the intermediate-frequency signal is shorter, and the delay time of the high-frequency signal is shortest. That is, the delay time of the high frequency first channel sample signal is less than the delay time of the intermediate frequency first channel sample signal, and the delay time of the intermediate frequency first channel sample signal is less than the delay time of the low frequency first channel sample signal.
In the weighting process, the weighting factor should be a value between 0 and 1, and by multiplying this value, the total volume is guaranteed to remain unchanged.
in the embodiment of the present invention, the second aspect of the processing of the left channel sample signals may be to weight the left channel sample signals that are not processed by the band pass filter, and the purpose of the second aspect of the processing of the left channel sample signals is to ensure the sound quality of the sound and maintain the original characteristics of the sound.
In a specific implementation, a weighting coefficient may be determined first during weighting, and then the left channel sample point signal that is not processed by the band pass filter is multiplied by the corresponding weighting coefficient, so as to ensure that the total volume remains unchanged and the interference effect between signals is strongest, thereby obtaining a left channel weighted signal.
in the embodiment of the present invention, an inverter may be disposed in the television, and the right channel sampling point signal is inverted through the inverter to obtain an inverted right channel sampling point signal.
as shown in fig. 6, the processing of the inverted second channel sample signal is similar to the processing of the first channel sample signal. That is, the second channel sampling point signal that has passed through the band pass filter is delayed and weighted to obtain a plurality of second channel band pass signals, and the second channel sampling point signal that has not passed through the band pass filter is directly weighted to obtain a second channel weighted signal.
In a specific implementation, a plurality of frequency bands of the inverted second channel sampling point signal may be determined, then the inverted second channel sampling point signal is transmitted to a band pass filter of a corresponding frequency band, the delayed and weighted processing is performed on the inverted second channel sampling point signal respectively to obtain a plurality of second channel band pass signals, and the weighted processing is performed on the inverted second channel sampling point signal to obtain a second channel weighted signal.
In general, the signal of a single speaker is not different in the human ear hearing after being inverted. However, in the case of the dual speakers, after the signal of the right speaker is subjected to the phase inversion, the signal of the right speaker strongly interferes with the signal of the original left speaker in the space, so that a relatively significant phase difference and sound intensity difference are generated at different points in the space, and a wide sound field area is formed in the human brain.
and step 510, inputting the second channel weighted signal and a plurality of second channel band-pass signals to a second channel loudspeaker.
In the embodiment of the present invention, after obtaining the first channel weighting signal and the plurality of first channel band pass signals, the first channel weighting signal and the plurality of first channel band pass signals may be input to a first channel speaker and played through the first channel speaker. The obtained second channel weighted signal and the multi-channel second channel band-pass signal can be input to a second channel loudspeaker and played through the second channel loudspeaker, so that an obvious virtual surround sound effect can be obtained under the coordination of the first channel loudspeaker and the second channel loudspeaker.
In the embodiment of the invention, the input signals are divided into frames and then enter the left loudspeaker and the right loudspeaker respectively through different processes. The signals entering the left speaker include two types, one is a weighted left channel weighted signal, and the other is a multi-channel left channel band pass signal after being subjected to a band pass filter, delayed and weighted. The left channel weighted signal can ensure the sound quality of the sound and keep the original characteristics of the sound; the band-pass filter can select signals of a certain frequency band, then the signals are sent to the left channel loudspeaker after delay and weighting processing, and each path of left channel band-pass signal increases the complexity of the signals by selecting signal segments with different frequencies. The signal fed to the right channel speaker is similar to the signal fed to the left channel speaker except that an inverter is added to the front and the signal from the single speaker is inverted to be indistinguishable in the human ear. However, for the dual speakers, after the signal of the right channel speaker is subjected to the phase inversion, the signal of the right channel speaker and the signal of the original left channel speaker generate strong interference in the space, and generate relatively obvious phase difference and sound intensity difference at different points in the space, so that a wide sound field area is formed in the human brain.
It should be noted that, for simplicity of description, the method embodiments are described as a series of acts or combination of acts, but those skilled in the art will recognize that the present invention is not limited by the illustrated order of acts, as some steps may occur in other orders or concurrently in accordance with the embodiments of the present invention. Further, those skilled in the art will appreciate that the embodiments described in the specification are presently preferred and that no particular act is required to implement the invention.
Referring to fig. 7, a schematic block diagram of a signal processing apparatus according to an embodiment of the present invention is shown, where the apparatus 700 may be applied to a virtual surround sound playing device, and the apparatus 700 may specifically include the following modules:
a receiving module 701, configured to receive an input signal;
a framing module 702, configured to frame the signal to obtain a signal to be processed, where the signal to be processed includes a first channel sampling point signal and a second channel sampling point signal;
a first generating module 703, configured to generate a first channel to-be-input signal based on the first channel sampling point signal; and the number of the first and second groups,
a second generating module 704, configured to perform an inverse operation on the second channel sampling point signal, and generate a second channel to-be-input signal based on the second channel sampling point signal after the inverse operation;
a first input module 705, configured to input the first channel signal to be input into a corresponding first channel speaker;
and the second input module 706 is configured to input the signal to be input in the second channel into a corresponding speaker in the second channel.
In this embodiment of the present invention, the framing module 702 may specifically include the following sub-modules:
the frame length determining submodule is used for determining the frame length of the framed signal;
and the signal framing submodule is used for dividing the signal into a plurality of frames of signals to be processed according to the frame length.
In this embodiment of the present invention, the first channel signal to be input may include a first channel weighting signal and multiple first channel band pass signals, and the first generating module 703 may specifically include the following sub-modules:
a first determining sub-module, configured to determine a plurality of frequency bands of the first channel sampling point signal;
the first transmission submodule is used for transmitting the first sound channel sampling point signal to a band-pass filter of a corresponding frequency band;
the first delay weighting processing sub-module is used for respectively carrying out delay and weighting processing on the first sound channel sampling point signals so as to obtain a plurality of paths of first sound channel band-pass signals; and the number of the first and second groups,
and the first weighting submodule is used for weighting the first channel sample point signal to obtain a first channel weighted signal.
In this embodiment of the present invention, the first delay weighting processing sub-module may specifically include the following units:
a first determining unit for determining a delay time and a weighting coefficient of the first channel band pass signal;
a first delay unit, configured to add a plurality of leading zeros in the first channel sample signal according to the delay time;
and the first weighting unit is used for multiplying the first channel sampling point signal after delay processing by the weighting coefficient so as to keep the total volume of the first channel sampling point signal unchanged.
In this embodiment of the present invention, the plurality of frequency bands may include a high frequency, a middle frequency, and a low frequency, where a delay time of the high frequency first channel sampling signal is less than a delay time of the middle frequency first channel sampling signal, and a delay time of the middle frequency first channel sampling signal is less than a delay time of the low frequency first channel sampling signal.
In this embodiment of the present invention, the second generating module 704 may specifically include the following sub-modules:
and the negation submodule is used for negating the second channel sampling point signal through a preset phase inverter to obtain a second channel sampling point signal after phase inversion.
In this embodiment of the present invention, the second channel to-be-input signal may include a second channel weighted signal and multiple second channel bandpass signals, and the second generating module 704 may further include the following sub-modules:
the second determining submodule is used for determining a plurality of frequency bands of the second channel sampling point signals after phase inversion;
the second transmission submodule is used for transmitting the second channel sampling point signal after phase inversion to a band-pass filter of a corresponding frequency band;
the second delay weighting processing submodule is used for respectively carrying out delay and weighting processing on the second channel sampling point signals after phase inversion so as to obtain a plurality of paths of second channel on-band signals; and the number of the first and second groups,
and the second weighting submodule is used for weighting the second channel sampling point signal after phase inversion so as to obtain a second channel weighting signal.
In the embodiment of the present invention, if the first channel is a left channel, the second channel is a right channel; if the first channel is a right channel, the second channel is a left channel.
Referring to fig. 8, there is shown a schematic block diagram of a virtual surround sound playing device according to an embodiment of the present invention, which may include a first channel speaker 801, a second channel speaker 802, and a signal processing apparatus 700 in the above embodiment.
For the apparatus embodiment and the apparatus embodiment, since they are substantially similar to the method embodiment, the description is relatively simple, and reference may be made to some descriptions of the method embodiment for relevant points.
The embodiments in the present specification are described in a progressive manner, each embodiment focuses on differences from other embodiments, and the same and similar parts among the embodiments are referred to each other.
As will be appreciated by one skilled in the art, embodiments of the present invention may be provided as a method, apparatus, or computer program product. Accordingly, embodiments of the present invention may take the form of an entirely hardware embodiment, an entirely software embodiment or an embodiment combining software and hardware aspects. Furthermore, embodiments of the present invention may take the form of a computer program product embodied on one or more computer-usable storage media (including, but not limited to, disk storage, CD-ROM, optical storage, and the like) having computer-usable program code embodied therein.
Embodiments of the present invention are described with reference to flowchart illustrations and/or block diagrams of methods, terminal devices (systems), and computer program products according to embodiments of the invention. It will be understood that each flow and/or block of the flow diagrams and/or block diagrams, and combinations of flows and/or blocks in the flow diagrams and/or block diagrams, can be implemented by computer program instructions. These computer program instructions may be provided to a processor of a general purpose computer, special purpose computer, embedded processor, or other programmable data processing terminal to produce a machine, such that the instructions, which execute via the processor of the computer or other programmable data processing terminal, create means for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.
These computer program instructions may also be stored in a computer-readable memory that can direct a computer or other programmable data processing terminal to function in a particular manner, such that the instructions stored in the computer-readable memory produce an article of manufacture including instruction means which implement the function specified in the flowchart flow or flows and/or block diagram block or blocks.
These computer program instructions may also be loaded onto a computer or other programmable data processing terminal to cause a series of operational steps to be performed on the computer or other programmable terminal to produce a computer implemented process such that the instructions which execute on the computer or other programmable terminal provide steps for implementing the functions specified in the flowchart flow or flows and/or block diagram block or blocks.
While preferred embodiments of the present invention have been described, additional variations and modifications of these embodiments may occur to those skilled in the art once they learn of the basic inventive concepts. Therefore, it is intended that the appended claims be interpreted as including preferred embodiments and all such alterations and modifications as fall within the scope of the embodiments of the invention.
Finally, it should also be noted that, herein, relational terms such as first and second, and the like may be used solely to distinguish one entity or action from another entity or action without necessarily requiring or implying any actual such relationship or order between such entities or actions. Also, the terms "comprises," "comprising," or any other variation thereof, are intended to cover a non-exclusive inclusion, such that a process, method, article, or terminal that comprises a list of elements does not include only those elements but may include other elements not expressly listed or inherent to such process, method, article, or terminal. Without further limitation, an element defined by the phrase "comprising an … …" does not exclude the presence of other like elements in a process, method, article, or terminal that comprises the element.
The foregoing detailed description of a signal processing method, a signal processing apparatus and a virtual surround sound playing device provided by the present invention, and the present disclosure describes the principle and implementation of the present invention by applying specific examples, and the description of the foregoing examples is only used to help understanding the method and the core idea of the present invention; meanwhile, for a person skilled in the art, according to the idea of the present invention, there may be variations in the specific embodiments and the application scope, and in summary, the content of the present specification should not be construed as a limitation to the present invention.
Claims (9)
1. A method of signal processing, applied to a virtual surround sound playing device, the method comprising:
receiving an input signal;
framing the signal to obtain a signal to be processed, wherein the signal to be processed comprises a first sound channel sampling point signal and a second sound channel sampling point signal;
generating a first channel to-be-input signal based on the first channel sampling point signal; and the number of the first and second groups,
carrying out phase inversion operation on the second channel sampling point signal, and generating a second channel signal to be input based on the second channel sampling point signal after phase inversion;
inputting the first channel signal to be input to a first channel loudspeaker;
inputting the second channel to-be-input signal to a second channel loudspeaker;
the first channel to-be-input signal includes a first channel weighting signal and multiple first channel band-pass signals, and the step of generating the first channel to-be-input signal based on the first channel sampling point signal includes:
determining a plurality of frequency bands of the first channel sampling point signal;
transmitting the first sound channel sampling point signal to a band-pass filter of a corresponding frequency band;
respectively carrying out delay and weighting processing on the first sound channel sampling point signals to obtain a plurality of paths of first sound channel band-pass signals; and the number of the first and second groups,
and carrying out weighting processing on the first channel sample point signal to obtain a first channel weighted signal.
2. The method according to claim 1, wherein the step of framing the signal to obtain the signal to be processed comprises:
determining the frame length of the framed signal;
and dividing the signal into a plurality of frames of signals to be processed according to the frame length.
3. The method according to claim 1, wherein the step of delaying and weighting the first channel sample signals respectively to obtain a plurality of first channel band pass signals comprises:
determining a delay time and a weighting coefficient of the first channel band-pass signal;
adding a plurality of leading zeros in the first channel sample point signal according to the delay time;
and multiplying the first channel sampling point signal after delay processing by the weighting coefficient to keep the total volume of the first channel sampling point signal unchanged.
4. The method of claim 3, wherein the plurality of frequency bands comprise a high frequency, a mid frequency, and a low frequency, wherein a delay time of the high frequency first channel sample signal is less than a delay time of the mid frequency first channel sample signal, and wherein a delay time of the mid frequency first channel sample signal is less than a delay time of the low frequency first channel sample signal.
5. The method of claim 1, 2, 3 or 4, wherein inverting the second channel sample signal comprises:
and inverting the second channel sampling point signal through a preset inverter to obtain an inverted second channel sampling point signal.
6. The method according to claim 5, wherein the second channel signal to be input includes a second channel weighted signal and a plurality of second channel band-pass signals, and the step of generating the second channel signal to be input based on the inverted second channel sample signal includes:
determining a plurality of frequency bands of the second channel sampling point signal after phase inversion;
transmitting the second channel sampling point signal after phase inversion to a band-pass filter of a corresponding frequency band;
respectively carrying out delay and weighting processing on the second channel sampling point signals after phase inversion to obtain a plurality of paths of second channel band-pass signals; and the number of the first and second groups,
and carrying out weighting processing on the second channel sampling point signal after phase inversion to obtain a second channel weighted signal.
7. The method of claim 6, wherein if the first channel is a left channel, the second channel is a right channel; if the first channel is a right channel, the second channel is a left channel.
8. An apparatus for signal processing, applied to a virtual surround sound playing device, the apparatus comprising:
the receiving module is used for receiving an input signal;
the framing module is used for framing the signal to obtain a signal to be processed, wherein the signal to be processed comprises a first sound channel sampling point signal and a second sound channel sampling point signal;
a first generating module, configured to generate a first channel to-be-input signal based on the first channel sampling point signal; and the number of the first and second groups,
the second generation module is used for carrying out phase inversion operation on the second channel sampling point signal and generating a second channel signal to be input based on the second channel sampling point signal after phase inversion;
the first input module is used for inputting the first channel to-be-input signal to a first channel loudspeaker;
and the second input module is used for inputting the signal to be input in the second channel to the loudspeaker in the second channel.
9. A virtual surround sound playing device comprising a first channel speaker, a second channel speaker, and the signal processing apparatus according to claim 8.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201810282945.5A CN108810737B (en) | 2018-04-02 | 2018-04-02 | Signal processing method and device and virtual surround sound playing equipment |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201810282945.5A CN108810737B (en) | 2018-04-02 | 2018-04-02 | Signal processing method and device and virtual surround sound playing equipment |
Publications (2)
| Publication Number | Publication Date |
|---|---|
| CN108810737A CN108810737A (en) | 2018-11-13 |
| CN108810737B true CN108810737B (en) | 2020-11-27 |
Family
ID=64095312
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201810282945.5A Active CN108810737B (en) | 2018-04-02 | 2018-04-02 | Signal processing method and device and virtual surround sound playing equipment |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN108810737B (en) |
Families Citing this family (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN109640243A (en) * | 2018-12-06 | 2019-04-16 | 广州番禺巨大汽车音响设备有限公司 | Virtual three-dimensional sound effect processing method and system based on the enhancing of DSP audio |
Citations (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101129091A (en) * | 2005-02-25 | 2008-02-20 | 雅马哈株式会社 | Array speaker apparatus |
| CN104053120A (en) * | 2014-06-13 | 2014-09-17 | 福建星网视易信息系统有限公司 | Method and device for processing stereo audio frequency |
| CN104010264B (en) * | 2013-02-21 | 2016-03-30 | 中兴通讯股份有限公司 | The method and apparatus of binaural audio signal process |
| CN106211007A (en) * | 2015-05-08 | 2016-12-07 | 深圳市三诺声智联股份有限公司 | A kind of stereo surrounding system of 3D |
-
2018
- 2018-04-02 CN CN201810282945.5A patent/CN108810737B/en active Active
Patent Citations (4)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101129091A (en) * | 2005-02-25 | 2008-02-20 | 雅马哈株式会社 | Array speaker apparatus |
| CN104010264B (en) * | 2013-02-21 | 2016-03-30 | 中兴通讯股份有限公司 | The method and apparatus of binaural audio signal process |
| CN104053120A (en) * | 2014-06-13 | 2014-09-17 | 福建星网视易信息系统有限公司 | Method and device for processing stereo audio frequency |
| CN106211007A (en) * | 2015-05-08 | 2016-12-07 | 深圳市三诺声智联股份有限公司 | A kind of stereo surrounding system of 3D |
Also Published As
| Publication number | Publication date |
|---|---|
| CN108810737A (en) | 2018-11-13 |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| JP5497217B2 (en) | Headphone correction system | |
| AU2014413559B2 (en) | A signal processing apparatus for enhancing a voice component within a multi-channel audio signal | |
| CA2835463C (en) | Apparatus and method for generating an output signal employing a decomposer | |
| JP6377249B2 (en) | Apparatus and method for enhancing an audio signal and sound enhancement system | |
| TW201727623A (en) | Apparatus and method for sound stage enhancement | |
| EP3046339A1 (en) | Virtual stereo synthesis method and device | |
| EP3005362B1 (en) | Apparatus and method for improving a perception of a sound signal | |
| JP2019083570A (en) | Method of crosstalk processing | |
| Romoli et al. | A mixed decorrelation approach for stereo acoustic echo cancellation based on the estimation of the fundamental frequency | |
| CN106797526A (en) | Apparatus for processing audio, methods and procedures | |
| KR20220080146A (en) | Subband spatial and crosstalk processing using spectrally orthogonal audio components | |
| US20200059750A1 (en) | Sound spatialization method | |
| CN114222226A (en) | Method, system, and medium for enhancing an audio signal having a left channel and a right channel | |
| KR102310859B1 (en) | Sound spatialization with room effect | |
| CN108810737B (en) | Signal processing method and device and virtual surround sound playing equipment | |
| US9794717B2 (en) | Audio signal processing apparatus and audio signal processing method | |
| WO2020036077A1 (en) | Signal processing device, signal processing method, and program | |
| US9641953B2 (en) | Sound spatialization with room effect, optimized in terms of complexity | |
| US10524052B2 (en) | Dominant sub-band determination | |
| KR20210021320A (en) | Perceptually-transparent estimation of a 2-channel spatial transfer function for sound correction | |
| CN112584300B (en) | Audio upmixing method, device, electronic equipment and storage medium | |
| Faller | Upmixing and beamforming in professional audio | |
| CN116320908A (en) | Virtual stereo generation method and electronic equipment | |
| AU2012252490A1 (en) | Apparatus and method for generating an output signal employing a decomposer |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| PB01 | Publication | ||
| PB01 | Publication | ||
| SE01 | Entry into force of request for substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| CB02 | Change of applicant information |
Address after: 266100, No. 218, Bay Road, Qingdao economic and Technological Development Zone, Shandong Applicant after: Hisense Visual Technology Co., Ltd. Address before: 266100, No. 218, Bay Road, Qingdao economic and Technological Development Zone, Shandong Applicant before: QINGDAO HISENSE ELECTRONICS Co.,Ltd. |
|
| CB02 | Change of applicant information | ||
| GR01 | Patent grant | ||
| GR01 | Patent grant |