CN106412763B - A kind of method and apparatus of audio processing - Google Patents
A kind of method and apparatus of audio processing Download PDFInfo
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- CN106412763B CN106412763B CN201610887914.3A CN201610887914A CN106412763B CN 106412763 B CN106412763 B CN 106412763B CN 201610887914 A CN201610887914 A CN 201610887914A CN 106412763 B CN106412763 B CN 106412763B
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- 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
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- 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
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Abstract
The invention discloses a kind of method and apparatus of audio processing.This method comprises: receiving audio signal, the audio signal is divided by high-frequency signal and low frequency signal according to preset division point；It determines delay parameter, and is delayed according to the delay parameter to the low frequency signal；Low frequency signal after the high-frequency signal and delay is mixed to get mixed frequency signal, and the mixed frequency signal is handled, so that the frequency response of each loudspeaker unit replay signal is consistent in loudspeaker array；By treated, mixed frequency signal is respectively sent in corresponding loudspeaker unit, so that the mixed frequency signal after loudspeaker unit reproduction process.Using the above method, when solving existing portable projector played file, user receives the nonsynchronous technical problem in audio and video direction.
Description
Technical field
The present invention relates to multimedia technology field more particularly to a kind of method and apparatus of audio processing.
Background technique
Projector can project to the vision signal received or picture signal on curtain, so that user clearly sees
See the image of projection, therefore, projector is widely used in family, office, school and public place of entertainment.It is especially portable
The appearance of formula projector makes projector's exquisiteness, portability, microminiaturization, entertainment orientation and the functionization that tradition is huge project
Technology more closeness to life and amusement.
In general, be also equipped with audio frequency broadcast system in portable projector, when portable projector plays video, using matching
The audio frequency broadcast system set plays corresponding audio signal.
However, user watches video image when watching the video file that portable projector plays, through front curtain,
The audio file that broadcasting is listened to by the portable projector in non-front, be easy to produce image harmony cent from experience, i.e. sound
Sound and image are asynchronous on direction.In order to avoid above-mentioned phenomenon, it will usually which selection is in curtain end external loudspeaker, to guarantee sound
Sound is synchronous on image direction, but will increase additional expense.
Summary of the invention
In view of this, the embodiment of the present invention provides a kind of method and apparatus of audio processing, it is existing portable to solve
When projector played file, user receives the nonsynchronous technical problem in audio and video direction.
In a first aspect, the embodiment of the invention provides a kind of methods of audio processing, comprising:
Audio signal is received, the audio signal is divided by high-frequency signal and low frequency signal according to preset division point；
It determines delay parameter, and is delayed according to the delay parameter to the low frequency signal；
Low frequency signal after the high-frequency signal and delay is mixed to get mixed frequency signal, and the mixed frequency signal is carried out
Processing, so that the frequency response of each loudspeaker unit replay signal is consistent in loudspeaker array；
By treated, mixed frequency signal is respectively sent in corresponding loudspeaker unit, so that loudspeaker unit reproduction process
Mixed frequency signal afterwards.
Second aspect, the embodiment of the invention also provides a kind of devices of audio processing, comprising:
Frequency division module, for receiving audio signal, according to preset division point by the audio signal be divided into high-frequency signal and
Low frequency signal；
Time delay module is delayed to the low frequency signal for determining delay parameter, and according to the delay parameter；
Processing module, for the low frequency signal after the high-frequency signal and delay to be mixed to get mixed frequency signal, and to institute
It states mixed frequency signal to be handled, so that the frequency response of each loudspeaker unit replay signal is consistent in loudspeaker array；
Playback module, for will treated that mixed frequency signal is respectively sent in corresponding loudspeaker unit so that loudspeaking
Mixed frequency signal after device unit reproduction process.
The method and apparatus of a kind of audio processing provided in an embodiment of the present invention, by preset division point by received audio
Signal is divided into high-frequency signal and low frequency signal, is delayed according to delay parameter to low frequency signal, and by high-frequency signal and delay
It is handled after low frequency signal mixing afterwards, by treated, mixed frequency signal is respectively sent in corresponding loudspeaker unit, with
The technological means of mixed frequency signal after making loudspeaker unit reproduction process solves each loudspeaker unit in loudspeaker array and resets
Acoustic characteristic inconsistence problems keep each loudspeaker unit phonation characteristics consistent by the method for signal processing, so that reset
Signal directive property is stronger, effectively inhibits the signal of non-master sound beam direction, conducive to the clear uppick reflective surface of user
It is synchronous with sound to realize mobile terminal image when projection plays video for audio signal.
Detailed description of the invention
By reading a detailed description of non-restrictive embodiments in the light of the attached drawings below, of the invention other
Feature, objects and advantages will become more apparent upon:
Fig. 1 is a kind of flow chart of the method for audio processing that the embodiment of the present invention one provides；
Fig. 2 is the loudspeaker array structural schematic diagram that the embodiment of the present invention one provides；
Fig. 3 is a kind of flow chart of the method for audio processing provided by Embodiment 2 of the present invention；
Fig. 4 is a kind of flow chart of audio-frequency processing method provided by Embodiment 2 of the present invention；
Fig. 5 is a kind of flow chart of the method for determining filter factor provided by Embodiment 2 of the present invention；
Fig. 6 is a kind of flow chart of the method for audio processing that the embodiment of the present invention three provides；
Fig. 7 is that the mobile terminal that the embodiment of the present invention three provides uses schematic diagram；
Fig. 8 a is the time-domain pulse response figure of the first loudspeaker unit 7111 measured in anechoic room；
Fig. 8 b is the time-domain pulse response figure of the second loudspeaker unit 7112 measured in anechoic room；
Fig. 8 c is the time-domain pulse response figure of the third loudspeaker unit 7113 measured in anechoic room；
Fig. 8 d is the time-domain pulse response figure of the 4th loudspeaker unit 7114 measured in anechoic room；
Fig. 9 is the target frequency response diagram determined using the method for regularization；
Figure 10 a is first filter filter factor waveform diagram；
Figure 10 b is second filter filter factor waveform diagram；
Figure 10 c is third filter filtering coefficient waveform diagram；
Figure 10 d is the 4th filter filtering coefficient waveform diagram；
Figure 11 is the flow diagram of Audio Signal Processing；
Figure 12 is that the mobile terminal that the embodiment of the present invention three provides uses schematic diagram；
Figure 13 a is the time-domain pulse response figure that the first loudspeaker unit 1212 is measured in anechoic room；
Figure 13 b is the time-domain pulse response figure that the second loudspeaker unit 1213 is measured in anechoic room；
Figure 13 c is the time-domain pulse response figure that third loudspeaker unit 1214 is measured in anechoic room；
Figure 13 d is the time-domain pulse response figure that the 4th loudspeaker unit 1215 is measured in anechoic room；
Figure 13 e is the time-domain pulse response figure that the 5th loudspeaker unit 1216 is measured in anechoic room；
Figure 14 a is first filter filter factor waveform diagram；
Figure 14 b is second filter filter factor waveform diagram；
Figure 14 c is third filter filtering coefficient waveform diagram；
Figure 14 d is the 4th filter filtering coefficient waveform diagram；
Figure 14 e is the 5th filter filtering coefficient waveform diagram；
Figure 15 is a kind of flow chart of the method for audio processing that the embodiment of the present invention four provides；
Figure 16 is that the mobile terminal that the embodiment of the present invention four provides uses schematic diagram；
Figure 17 a is the time-domain pulse response figure that the first loudspeaker unit 1612 is measured in anechoic room；
Figure 17 b is the time-domain pulse response figure that the second loudspeaker unit 1613 is measured in anechoic room；
Figure 17 c is the time-domain pulse response figure that third loudspeaker unit 1614 is measured in anechoic room；
Figure 17 d is the time-domain pulse response figure that the 4th loudspeaker unit 1615 is measured in anechoic room；
Figure 17 e is the time-domain pulse response figure that the 5th loudspeaker unit 1616 is measured in anechoic room；
Figure 17 f is the time-domain pulse response figure that the 6th loudspeaker unit 1617 is measured in anechoic room；
Figure 18 is the target frequency response diagram determined using the method for regularization；
Figure 19 a is the first filter filter factor waveform diagram for handling the first mixed frequency signal；
Figure 19 b is the second filter filter factor waveform diagram for handling the first mixed frequency signal；
Figure 19 c is the third filter filtering coefficient waveform diagram for handling the first mixed frequency signal；
Figure 19 d is the 4th filter filtering coefficient waveform diagram for handling the first mixed frequency signal；
Figure 19 e is the 5th filter filtering coefficient waveform diagram for handling the first mixed frequency signal；
Figure 19 f is the 6th filter filtering coefficient waveform diagram for handling the first mixed frequency signal；
Figure 20 a is the first filter filter factor waveform diagram for handling the second mixed frequency signal；
Figure 20 b is the second filter filter factor waveform diagram for handling the second mixed frequency signal；
Figure 20 c is the third filter filtering coefficient waveform diagram for handling the second mixed frequency signal；
Figure 20 d is the 4th filter filtering coefficient waveform diagram for handling the second mixed frequency signal；
Figure 20 e is the 5th filter filtering coefficient waveform diagram for handling the second mixed frequency signal；
Figure 20 f is the 6th filter filtering coefficient waveform diagram for handling the second mixed frequency signal；
Figure 21 is the flow diagram of double-audio signal processing；
Figure 22 is the structural schematic diagram of the device for the audio processing that the embodiment of the present invention five provides.
Specific embodiment
The present invention is described in further detail with reference to the accompanying drawings and examples.It is understood that this place is retouched
The specific embodiment stated is used only for explaining the present invention rather than limiting the invention.It also should be noted that in order to just
In description, only some but not all contents related to the present invention are shown in the drawings.
Embodiment one
Fig. 1 is a kind of flow chart of the method for audio processing that the embodiment of the present invention one provides.Side provided in this embodiment
Method is suitable for carrying out the situation that video projection plays using Portable projector.Method provided in this embodiment can be by audio
The device of reason executes, which can be realized by way of software and/or hardware, and be integrated in the mobile terminal of projectable
In.With reference to Fig. 1, method provided in this embodiment is specifically included:
S110, audio signal is received, the audio signal is divided by high-frequency signal and low frequency signal according to preset division point.
Specifically, mobile terminal is followed by receiving audio signal in unlatching projection.In the present embodiment, mobile terminal is preferably just
Formula projector is taken, which passes through loudspeaker array playback audio signal.Fig. 2 is what the embodiment of the present invention one provided
Loudspeaker array structural schematic diagram, with reference to Fig. 2, loudspeaker array 1 includes multiple loudspeaker units 11, each 11 line of loudspeaker unit
Property arrangement, and the spacing between each loudspeaker unit 11 be D.Letter compared with high directivity can be reset using Loudspeaker line combination
Number, reflecting surface is reached convenient for signal and is reflected.
In view of including low frequency signal and high-frequency signal in the received audio signal of mobile terminal, and loudspeaker array is in weight
When putting audio signal, high-frequency signal directive property with higher can reach reflecting surface and be reflected according to preset direction, and
The directive property of low frequency signal is very weak, playback time propagation without directivity, so, it may appear that the low frequency signal of synchronization is than high
The case where frequency signal is reached at user earlier, so that the audio distortions that user hears.The generation of above situation in order to prevent, in advance
Part-frequency point is first set, audio signal is divided into high-frequency signal and low frequency signal, and further process to low frequency signal, so that
The high-frequency signal of synchronization reaches at user's ears earlier than treated low frequency signal, and the low frequency signal meeting that guarantees that treated
It being reached at user's ears in 30ms after high-frequency signal reaches at user's ears, wherein 30ms is critical value, after 30ms,
User hears the audio signal of distortion.
Wherein, preset division point can be set according to the length of loudspeaker array.The length of loudspeaker array is to raise
Total length in sound device array after loudspeaker unit arrangement, the spacing between each loudspeaker unit can carry out according to the actual situation
Setting.
S120, it determines delay parameter, and is delayed according to the delay parameter to the low frequency signal.
Specifically, carrying out delay process to low frequency signal, loudspeaker array is in playback audio signal, low frequency signal and height
There can be delay between frequency signal, so that the high-frequency signal after reflection reaches user prior to not reflected low frequency signal
Place.Wherein, delay parameter can be determined at a distance from reflecting surface by mobile terminal.
S130, the low frequency signal after the high-frequency signal and delay is mixed to get mixed frequency signal, and the mixing is believed
It number is handled, so that the frequency response of each loudspeaker unit replay signal is consistent in loudspeaker array.
Typically, when loudspeaker array playback audio signal, the characteristic coherency of each loudspeaker unit is higher, audio playback
The directive property of the acoustic beam generated when signal is higher.In the present embodiment, guarantee the feature of each loudspeaker unit to guarantee each loudspeaking
The frequency response of device unit replay signal is consistent, wherein the frequency response of replay signal is denoted as target frequency response.
Specifically, mixed frequency signal can be handled according to the frequency response of loudspeaker unit each in loudspeaker array,
To guarantee treated mixed frequency signal after loudspeaker unit is reset, frequency response is consistent (to ring in the present embodiment for target frequency
Answer), the problem of inconsistency of each loudspeaker unit of loudspeaker array can be corrected in this way, and utmostly inhibits loudspeaker battle array
The secondary lobe acoustic beam of column.Wherein, specifically processing rule can be set according to the actual situation.
Optionally, if the difference of frequency response of each loudspeaker unit is excessive, first mixed frequency signal can be carried out at branch
Reason respectively believes the mixing of branch according to the frequency response of each loudspeaker unit per a loudspeaker unit is corresponded to all the way
It number is handled.
S140, by treated, mixed frequency signal is respectively sent in corresponding loudspeaker unit, so that loudspeaker unit weight
Mixed frequency signal of putting that treated.
Specifically, resetting the mixed frequency signal after each loudspeaker unit receives that treated mixed frequency signal.Wherein it is possible to
It is to reset mixed frequency signal according to preset mixed frequency signal propagation angle.For example, angle of radiation is 90 °.In the mixed frequency signal of playback
High-frequency signal reaches at user after the reflection of reflecting surface, and the low frequency signal after delay is later than high-frequency signal and reaches at user.
Optionally, reflecting surface can be metope, can save the flower for additionally purchasing reflecting surface when user uses mobile terminal in this way
Pin.
Received audio signal is divided into high frequency letter by preset division point by the technical solution that the embodiment of the present invention one provides
Number and low frequency signal, be delayed according to delay parameter to low frequency signal, and by high-frequency signal and delay after low frequency signal mix
It is handled after conjunction, by treated, mixed frequency signal is respectively sent in corresponding loudspeaker unit, so that loudspeaker unit weight
The technological means for mixed frequency signal of putting that treated, realizes the frequency response of each loudspeaker unit replay signal in loudspeaker array
Unanimously, that is, the acoustic characteristic for the signal reset is consistent, so that the signal directive property reset is stronger, effectively inhibits non-master sound
The signal of Shu Fangxiang realizes mobile terminal and broadcasts in projection conducive to the audio signal of the clear uppick reflective surface of user
Image is synchronous with sound when putting video.
Embodiment two
Fig. 3 is a kind of flow chart of the method for audio processing provided by Embodiment 2 of the present invention.The present embodiment is above-mentioned
It is optimized on the basis of embodiment.With reference to Fig. 3, method provided in this embodiment is specifically included:
S210, audio signal is received, the audio signal is divided by high-frequency signal and low frequency signal according to preset division point.
S220, reflective distance parameter is obtained, and determines that the delay of the low frequency signal is joined according to the reflective distance parameter
Number, and be delayed according to the delay parameter to the low frequency signal.
Wherein, reflective distance parameter is loudspeaker array at a distance from reflecting surface.
It, can input reflection in the terminal in advance specifically, user is when carrying out projection using mobile terminal and playing
Distance parameter.Mobile terminal determines delay parameter according to reflective distance parameter.
It further, can be according to formulaDetermine the delay parameter of the low frequency signal.Its
In, PFFor the reflective distance parameter of acquisition, Δ t is preset time parameter.Preferably, the range of Δ t is 1ms to 30ms,
Specific numerical value can be selected according to the actual situation.Determine PFAfter Δ t, delay parameter can be determined.Using the party
The delay parameter that method determines, it is ensured that after the high-frequency signal of same playing time reaches at user's ears, in 2ms after delay
Low frequency signal reaches at user's ears.
Specifically, delay parameter and preset sample frequency are multiplied to obtain discrete adopt when being delayed to low frequency signal
Sampling point delay is delayed to low frequency signal according to sampled point delay.Wherein, preset sample frequency is that mobile terminal believes low frequency
Number sampling frequency, can be determined according to the actual situation.
S230, the low frequency signal after the high-frequency signal and delay is mixed to get mixed frequency signal, to the mixed frequency signal
Carry out branch.
Wherein, the number of branch is identical as the number of loudspeaker unit in loudspeaker array.Specifically, due to each loudspeaker
The frequency response of unit may be not quite identical, therefore, carries out branch, and a corresponding loudspeaking per signal all the way to mixed frequency signal
Device unit, it is subsequent that every road mixed frequency signal is handled respectively, to guarantee the letter of the loudspeaker unit playback of different frequency response
Number frequency response it is consistent.
S240, every road mixed frequency signal is handled respectively.
In the present embodiment, the mode for handling mixed frequency signal is the side being filtered using filter to every road mixed frequency signal
Formula.Wherein, a corresponding filter per mixed frequency signal all the way.Specifically, Fig. 4 is a kind of audio provided by Embodiment 2 of the present invention
The flow chart of processing method, with reference to Fig. 4, which may include；
S241, the filter factor for determining the corresponding filter of each loudspeaker unit.
Specifically, the filter factor of filter is determined according to the frequency response of each loudspeaker unit, so that filter root
After filtering according to filter factor to mixed frequency signal, the frequency response of mixed frequency signal is consistent after each loudspeaker unit playback filtering.
Further, Fig. 5 is a kind of flow chart of the method for determining filter factor provided by Embodiment 2 of the present invention.With reference to
Fig. 5, method provided in this embodiment specifically include:
S2411, the frequency response for obtaining each loudspeaker unit, the target frequency response of loudspeaker array and loudspeaker
The Main beam angle of array transmission audio signal.
Specifically, the time domain response of each loudspeaker unit in loudspeaker array is measured under the conditions of anechoic room in advance.It surveys
After measuring time domain response, Fourier transformation is carried out to each time domain response and obtains the corresponding frequency response of each loudspeaker unit.Wherein,
Influence of other audio signals to measurement result can be eliminated under the conditions of anechoic room, so that measurement result is more accurate.
Further, determine that the target frequency of loudspeaker array replay signal responds using the method for regularization.Wherein, no
With requirement when loudspeaker array specification it is different, determine that target frequency response is different, wherein the specification of loudspeaker array
It may include the size of loudspeaker unit and the spacing of each loudspeaker unit.Target when using for example, family uses and handles official business
Frequency response can be different.Benefit using regularization is the frequency response of determining target frequency response and each loudspeaker unit
Difference will not be excessive, to the damage of hardware and subtract when can reduce audio signal of each loudspeaker unit after reproduction process
Few non-linear distortion.
Specifically, first setting the Main beam angle of loudspeaker array replay signal.Main beam angle is loudspeaker array weight
When putting audio signal, the main direction of propagation of audio signal and the angle of loudspeaker array long axis.Mobile terminal projected position is different
When, Main beam angle is different.User can be inputted desired in the terminal in advance when being projected using mobile terminal
Main beam angle.
S2412, according to each frequency response, the target frequency response and the Main beam angle, utilize minimum norm
Least square solution determines the frequency domain filtering coefficient of each filter.
Wherein, the formula of LS solution of the least norm are as follows: minimum (‖ CX-B ‖2)=0, whereinWherein, loudspeaker array is total
Include N number of loudspeaker unit, ΩmFor loudspeaker array send audio signal m-th of acoustic beam angle, m=(1,2 ...,
Tar ..., M), ΩtarFor the Main beam angle of the loudspeaker array playback audio signal of acquisition, remaining is in addition to Main beam
Audio signal angle of radiation, HnFor the frequency response of n-th of loudspeaker unit, n=(1,2 ..., N).It is n-th
Transfer function of the loudspeaker unit under m-th of acoustic beam angle,Spacing between loudspeaker unit, of loudspeaker unit
Several and Main beam angle is related.After user inputs Main beam angle, mobile terminal can be according to Main beam angle, loudspeaker
The number of spacing and loudspeaker unit between unit determinesOccurrence.XnFor the corresponding filter of n-th of loudspeaker unit
The frequency domain filtering coefficient of wave device, H0For the target frequency response of the loudspeaker array of acquisition, B (Ωm) it is ΩmUpper audio signal
Acoustic pressure.Preset the acoustic pressure B (Ω of audio signal in Main beam angletar) it is 1, the sound of audio signal in remaining acoustic beam angle
Pressure is 0, after the frequency domain filtering coefficient for the filter being calculated can in this way filtered mixed frequency signal, each loudspeaker unit weight
When putting filtered mixed frequency signal, which concentrates on Main beam angular spread.K is the loudspeaker battle array of setting
The discrete value of the audible frequency of column,F is the audible frequency of loudspeaker array, and value can be according to the actual situation
It is set, the f of such as 2 cun of loudspeakers can be set to about 160Hz-20kHz, and c is constant.
Using above-mentioned formula, the filter factor on each loudspeaker frequency domain can be determined.
S2413, using inversefouriertransform, be corresponding time domain filter coefficients as respectively using each frequency domain filtering transformation of coefficient
The filter factor of filter.
The filter factor of each filter obtained using LS solution of the least norm is numerical value on frequency domain, but filter is logical
Signal processing is often carried out in a manner of convolution filtering, therefore utilizes inversefouriertransform, by each frequency domain filtering transformation of coefficient
Filter factor for corresponding time domain filter coefficients as each filter.
S242, every road mixed frequency signal is respectively sent in corresponding filter, so that filter of each filter according to itself
Wave system number is filtered.
Specifically, each filter is filtered mixed frequency signal according to the filter factor of itself.
S250, filtered mixed frequency signal is respectively sent in corresponding loudspeaker unit, so that loudspeaker unit weight
Put filtered mixed frequency signal.
Received audio signal is divided into high-frequency signal by preset division point and low frequency is believed by the technical solution of the present embodiment
Number, computation delay parameter, and be delayed according to delay parameter to low frequency signal, it is determined using LS solution of the least norm each
The filter factor of the corresponding filter of loudspeaker unit is sent to each filter after mixing the low frequency signal after high-frequency signal and delay
Wave device is filtered, and filtered mixed frequency signal is sent to corresponding loudspeaker unit, so that loudspeaker unit weight
The technological means for mixed frequency signal of putting that treated, the frequency for realizing the signal that each loudspeaker unit is reset in loudspeaker array are rung
Should unanimously, that is, the acoustic characteristic for the signal reset is consistent, so that the signal directive property reset is stronger, is effectively inhibited non-master
The signal of sound beam direction realizes mobile terminal and is projecting conducive to the audio signal of the clear uppick reflective surface of user
Image is synchronous with sound when playing video.
Embodiment three
Fig. 6 is a kind of flow chart of the method for audio processing that the embodiment of the present invention three provides.The present embodiment is in above-mentioned reality
It applies and optimizes on the basis of example.With reference to Fig. 6, method provided in this embodiment is specifically included:
S310, audio signal is received, the audio signal is divided by high-frequency signal and low frequency signal according to preset division point.
S320, the low frequency signal is divided by low frequency subsignal and bass signal according to default bass part-frequency point.
When in view of using loudspeaker array, the low-frequency cut-off frequency of loudspeaker unit is relatively low.Accordingly, in mobile terminal
One bass effect unit of middle setting, to make up the low frequency missing of loudspeaker unit.Wherein, it is mobile whole for presetting bass part-frequency point
The cutoff frequency for the high-pass filter being arranged in end, or the cutoff frequency for the low-pass filter being arranged in mobile terminal,
The design parameter of high-pass filter and low-pass filter can be according to the feature of loudspeaker unit and the application scenarios of mobile terminal
(such as home scenarios or office scene) determine.
S330, it determines delay parameter, and low frequency subsignal and bass signal is prolonged respectively according to the delay parameter
When.
S340, the low frequency subsignal after the high-frequency signal and delay is mixed to get mixed frequency signal, and to the mixing
Signal is handled, so that the frequency response of each loudspeaker unit replay signal is consistent in loudspeaker array.
S350, will treated that mixed frequency signal is respectively sent in corresponding loudspeaker unit, and by the bass after delay
Signal is sent to bass effect unit, so that the bass effect unit resets the bass signal after the delay.
Illustratively illustrate method provided in this embodiment below, wherein the processing mode of mixed frequency signal is to utilize filtering
Device is filtered mixed frequency signal.
Example one, Fig. 7 are that the mobile terminal that the embodiment of the present invention three provides uses schematic diagram.With reference to Fig. 7, mobile terminal 71
Including loudspeaker array 711, loudspeaker array 711 is by 4 loudspeaker units (7111-7114 in Fig. 7) and 1 bass effect list
Member 712 forms, and the position where user 72 determines Main beam angle ΩtarIt is 90 °, the audio that loudspeaker array 711 is reset
Signal is reflected by wall 73.Further, the size of each loudspeaker unit be 27mm, low-frequency cut-off frequency 208Hz, i.e.,
Default bass part-frequency point is 208Hz, and preset division point is 1500Hz, preset sample frequency 48kHz, and loudspeaker unit spacing is
30mm, reflective distance parameter PFIt is 5ms, accordingly, determining delay parameter t for 3m, Δ tdFor 0.0226s, and then obtain discrete
Sampled point delay is 1085 points.Fig. 8 a- Fig. 8 d is respectively that the first loudspeaker unit 7111 to the 4th measured in anechoic room is raised
The time-domain pulse response figure of sound device unit 7114.Fig. 9 is the target frequency response diagram determined using the method for regularization.Due to
Ωtar=90 °, then B (Ωtar)=1, B (Ωi)=0 |i≠90°.It further, can be with according to LS solution of the least norm formula
The filter factor of corresponding 4 filters of 7111 to the 4th loudspeaker unit of the first loudspeaker unit 7114 is determined respectively.Figure
10a- Figure 10 d is each filter filtering coefficient waveform diagram.Figure 11 is the flow diagram of Audio Signal Processing.With reference to figure
11, the received audio signal of mobile terminal is divided by classifier 111, wherein frequency divider 111 is according to preset division point and presets
Audio signal is divided into high-frequency signal, low frequency subsignal and bass signal by bass part-frequency point, the frequency dividing of frequency divider 111 is obtained low
Frequency subsignal and bass signal are respectively sent to the first delay unit 1121 and the second delay unit 1122, the first delay unit
Low frequency subsignal after 1121 delays is sent in filter group 113 after mixing with high-frequency signal by frequency mixer 114, wherein
Mixed frequency signal can be divided into 4 tunnels and be respectively sent to first filter 1131, second filter 1132,1133 and of third filter
In 4th filter 1134.Each filter is filtered mixed frequency signal according to respective filter factor.By first filter
1131 filtered mixed frequency signals are sent to the first loudspeaker unit 7111, by the filtered mixed frequency signal of second filter 1132
It is sent to the second loudspeaker unit 7112, the filtered mixed frequency signal of third filter 1133 is sent to third loudspeaker unit
7113, the filtered mixed frequency signal of the 4th filter 1134 is sent to the 4th loudspeaker unit 7114, and the second delay is single
Bass signal after 1122 delay of member is sent to bass effect unit 712.Each loudspeaker unit is reset corresponding filtered mixed
Frequency signal, bass effect unit reset the bass signal after delay.
Example two, Figure 12 are that the mobile terminal that the embodiment of the present invention three provides uses schematic diagram.With reference to Figure 12, mobile terminal
121 include loudspeaker array 1211, and loudspeaker array 1211 is by 5 loudspeaker units (being respectively 1212-1216 in Figure 12) and 1
A bass effect unit 1217 forms, and mobile terminal 121 not arrange in a room by heart line, and the position where user 122 is true
Determine Main beam angle ΩtarIt is 45 °, the audio signal that loudspeaker array 1211 is reset is reflected by wall 123.Further
, the size of each loudspeaker unit is 27mm, low-frequency cut-off frequency 208Hz, i.e., default bass part-frequency point is 208Hz, presets and divides
Frequency point is 1500Hz, and preset sample frequency 48kHz, loudspeaker unit spacing is 30mm, reflective distance parameter PFFor 3m, Δ t
For 5ms, accordingly, determining delay parameter tdFor 0.0226s, and then obtaining discrete sampled point delay is 1085 points.Figure 13 a-
Figure 13 e is respectively the time-domain pulse response that 1212 to the 5th loudspeaker unit 1216 of the first loudspeaker unit is measured in anechoic room
Figure.The specification of loudspeaker unit used in the specification of the loudspeaker unit used in this example and above-mentioned example is identical, therefore
The target frequency response sought is identical as the target frequency response sought in above-mentioned example.Due to Ωtar=45 °, then B (Ωtar)
=1, B (Ωi)=0 |i≠45°.Further, 5 filters pair can be determined according to LS solution of the least norm formula respectively
The filter factor answered.Figure 14 a- Figure 14 e is each filter filtering coefficient waveform diagram.The process of specific Audio Signal Processing
Similar to the process of the processing of example one, it is not described here in detail.
The technical solution of the present embodiment is arranged bass effect unit in mobile terminal, and low frequency signal is divided into low frequency
Signal and bass signal can make up the low frequency of loudspeaker unit using the bass signal after bass effect unit forward delay interval
Missing, promotes the usage experience of user.
Example IV
Figure 15 is a kind of flow chart of the method for audio processing that the embodiment of the present invention four provides.The present embodiment is with above-mentioned reality
It applies and optimizes based on example, in the present embodiment, received audio signal includes the first audio signal and the second audio signal,
And first corresponding from the second audio signal Main beam angle of audio signal it is different.With reference to Figure 15, method provided in this embodiment
It specifically includes:
S410, the first audio signal and the second audio signal are received, is divided into the first audio signal according to preset division o'clock
Second audio signal is divided into the second high-frequency signal and the second low frequency signal by the first high-frequency signal and the first low frequency signal.
Wherein, the first audio signal is identical with the preset division point of the second audio signal.
S420, delay parameter is determined, and according to the delay parameter respectively to the first low frequency signal and the second low frequency signal
It is delayed.
Wherein, due to same mobile terminal reflective distance parameter PFIt is identical with preset time parameter Δ t, so first is low
The delay parameter that frequency signal and the second low frequency signal determine is identical.
S430, the first low frequency signal after the first high-frequency signal and delay is mixed to get the first mixed frequency signal, and to the
One mixed frequency signal is handled, and the second low frequency signal after the second high-frequency signal and delay is mixed to get the second mixed frequency signal,
And the second mixed frequency signal is handled.
Specifically, when being handled using filtered method the first mixed frequency signal and the second mixed frequency signal, due to first
Audio signal is different with the Main beam angle of the second audio signal, is using what LS solution of the least norm sought each filter
When number, filter factor is different when corresponding the first mixed frequency signal of filter process of same loudspeaker unit and the second mixed frequency signal,
Therefore, each loudspeaker unit needs corresponding two filters, wherein first mixed frequency signal of filter process, another
The second mixed frequency signal of filter process.
S440, will be according to the first mixed frequency signal after first Audio Signal Processing and according to the second Audio Signal Processing
The second mixed frequency signal afterwards is superimposed and is respectively sent in corresponding loudspeaker unit.
Further, loudspeaker unit the first mixed frequency signal after reproduction process and treated the second mixing letter simultaneously
Number.
Optionally, if in mobile terminal further including bass effect unit, according to default bass part-frequency point respectively by first
Low frequency signal is divided into the first low frequency subsignal and the first bass signal, and the second low frequency signal is divided into the second low frequency subsignal and
Two bass signals, and be delayed delay unit is sent to after the first bass signal and the superposition of the second bass signal, it is subsequent
Processing is identical as the processing method that embodiment three provides, and is not detailed herein.
Illustratively illustrate method provided in this embodiment below.Wherein, the processing mode of mixed frequency signal is to utilize filtering
Device is filtered mixed frequency signal
Figure 16 is that the mobile terminal that the embodiment of the present invention four provides uses schematic diagram.With reference to Figure 16, mobile terminal 161 includes
Loudspeaker array 1611, loudspeaker array 1611 is by 6 loudspeaker units (being respectively 1612-1617 in Figure 16) and 1 bass
Effect unit 1618 forms, and mobile terminal 161 receives the first audio signal and the second audio signal, according to 162 place of user
Position determines the Main beam angle Ω of the first audio signal1tarIt is 30 °, the Main beam angle Ω of the second audio signal2tarFor
150 °, loudspeaker array 1611 resets the audio signal in both direction and carries out two secondary reflections by reflecting surface 163, at this time user
The audio signal of 162 uppick both directions, i.e. uppick solid sound.Further, the size of each loudspeaker unit is 52mm,
Low-frequency cut-off frequency 180Hz, i.e., default bass part-frequency point is 180Hz, and preset division point is 1500Hz, and preset sample frequency is
48kHz, loudspeaker unit spacing are 52mm, reflective distance parameter PFIt is 5ms, accordingly, determining delay parameter t for 3m, Δ tdFor
0.0226s, and then obtaining discrete sampled point delay is 1085 points.Figure 17 a- Figure 17 f is respectively that first is measured in anechoic room
The time-domain pulse response figure of 1612 to the 6th loudspeaker unit 1617 of loudspeaker unit.Figure 18 is to be determined using the method for regularization
Target frequency response diagram.When handling the first audio signal, due to Ω1tar=30 °, then B (Ω1tar)=1, B (Ω1i)=0
|1i≠30°, the corresponding filtering of filter of 6 the first mixed frequency signals of processing can be determined according to LS solution of the least norm formula
Coefficient.Figure 19 a- Figure 19 f is each filter filtering coefficient waveform diagram for handling the first mixed frequency signal.In the second audio signal
When, due to Ω2tar=150 °, then B (Ω2tar)=1, B (Ω2i)=0 |2i≠45°, it can according to LS solution of the least norm formula
To determine the corresponding filter factor of filter of 6 the second mixed frequency signals of processing.Figure 20 a- Figure 20 f is the second mixed frequency signal of processing
Each filter filtering coefficient waveform diagram.Figure 21 is the flow diagram of double-audio signal processing.With reference to Figure 21, wherein
First frequency divider 2111 divides the first audio signal to obtain the first low frequency subsignal, the first bass signal and the first high frequency letter
Number, the second frequency divider 2112 divides the second audio signal to obtain the second low frequency subsignal, the second bass signal and the second high frequency
Signal is sent to the second delay unit by the first bass signal and the second bass signal after the first frequency mixer 2151 mixes
2122, the bass signal of the second 2122 pairs of delay unit mixing is sent to bass effect unit 1618 after being delayed, by bass
Effect unit 1618 is reset outward.The process flow of first high-frequency signal and the first low frequency subsignal, the second high-frequency signal and
The process flow phase of high-frequency signal and low frequency signal that the process flow of two low frequency subsignals is provided with example one in embodiment three
Together.First delay unit 2121, the second delay unit 2122 are identical with the delay parameter of third delay unit 2223.It will be used to locate
Manage filtered first mixed frequency signal of first filter 2131 of the first mixed frequency signal and for handling the second mixed frequency signal
Filtered second mixed frequency signal of one filter 2141 is sent to loudspeaker unit 1612 simultaneously, is sent to and raises after can also being superimposed
Sound device unit 1612, so that first mixed frequency signal of the loudspeaker unit 1612 after 30 ° of direction reproduction process, in 150 ° of direction weights
Putting treated, the second mixed frequency signal is similarly respectively handled each the first mixed frequency signal of road and the second mixed frequency signal, so that
Loudspeaker unit 1613 is to the first mixed frequency signal after 30 ° of direction reproduction process respectively of loudspeaker unit 1617,150 ° of sides
To the second mixed frequency signal after reproduction process.
The technical solution of the present embodiment, when receiving the first audio signal and the second audio signal, by respectively to
The processing of one audio signal and the second audio signal realizes the audio letter that two Main beam angles are reset in loudspeaker array
Number, so that the sound played has stereophonic effect, improve the usage experience of user.
Embodiment five
Figure 22 is the structural schematic diagram of the device for the audio processing that the embodiment of the present invention five provides.With reference to Figure 22, this implementation
The device for the audio processing that example provides includes: frequency division module 2201, time delay module 2202, processing module 2203 and playback module
2204。
Wherein, the audio signal is divided into height according to preset division point for receiving audio signal by frequency division module 2201
Frequency signal and low frequency signal；Time delay module 2202 believes the low frequency for determining delay parameter, and according to the delay parameter
It number is delayed；Processing module 2203, for the low frequency signal after the high-frequency signal and delay to be mixed to get mixed frequency signal,
And the mixed frequency signal is handled, so that the frequency response of each loudspeaker unit replay signal is consistent in loudspeaker array；
Playback module 2204, for will treated that mixed frequency signal is respectively sent in corresponding loudspeaker unit so that loudspeaker list
Mixed frequency signal after first reproduction process.
On the basis of the above embodiments, the time delay module 2202 includes: parameter acquisition submodule, for obtaining reflection
Distance parameter, and determine according to the reflective distance parameter delay parameter of the low frequency signal, wherein the reflective distance ginseng
Number is loudspeaker array at a distance from reflecting surface；Be delayed submodule, for according to the delay parameter to the low frequency signal into
Line delay.
On the basis of the above embodiments, the parameter acquisition submodule is specifically used for: according to formula:Determine the delay parameter of the low frequency signal, wherein PFFor the reflective distance parameter of acquisition, Δ t
For preset time parameter, tdFor delay parameter.
On the basis of the above embodiments, the processing module 2203 includes: mixing submodule, for believing the high frequency
Number and delay after low frequency signal be mixed to get mixed frequency signal, branch submodule, for carrying out branch to the mixed frequency signal,
In, the number of branch is identical as the number of loudspeaker unit in loudspeaker array；Submodule is handled, for being mixed respectively to every road
Signal is handled.
On the basis of the above embodiments, the processing submodule includes: filter factor determination unit, each for determining
The filter factor of the corresponding filter of loudspeaker unit；Filter unit, it is corresponding for every road mixed frequency signal to be respectively sent to
In filter, so that each filter is filtered according to the filter factor of itself.
On the basis of the above embodiments, the filter factor determination unit includes: that numerical value obtains subelement, for obtaining
The frequency response of each loudspeaker unit, the target frequency response of loudspeaker array and loudspeaker array playback audio signal
Main beam angle；Frequency coefficient determines subelement, for according to each frequency response, target frequency response and the main sound
Beam angle degree determines the frequency domain filtering coefficient of each filter using LS solution of the least norm；Time-domain coefficients determine subelement, use
It is corresponding time domain filter coefficients as the filtering of each filter using each frequency domain filtering transformation of coefficient in utilizing inversefouriertransform
Coefficient.
On the basis of the above embodiments, the formula of the LS solution of the least norm are as follows: minimum (‖ CX-B
‖2)=0, whereinWherein, it raises
Sound device array includes N number of loudspeaker unit, Ω altogethermFor m-th of acoustic beam angle of loudspeaker array playback audio signal, m=(1,
2 ..., tar ..., M) wherein, ΩtarFor the Main beam angle of the loudspeaker array playback audio signal of acquisition, remaining is
Audio signal angle of radiation in addition to Main beam, HnFor the frequency response of n-th of loudspeaker unit, n=(1,2 ..., N),For transfer function of n-th of loudspeaker unit under m-th of acoustic beam angle, XnFor the corresponding filter of n-th of loudspeaker unit
The frequency domain filtering coefficient of wave device, H0For the target frequency response of the loudspeaker array of acquisition, B (Ωm) it is ΩmUpper audio signal
Acoustic pressure, k are the discrete value of the audible frequency of the loudspeaker array of setting.
On the basis of the above embodiments, further includes: low frequency frequency division module, for according to preset division point by the sound
Frequency signal is divided into after high-frequency signal and low frequency signal, and the low frequency signal is divided into low frequency letter according to default bass part-frequency point
Number and bass signal.
Correspondingly, the processing module 2203 is specifically used for: the low frequency subsignal after the high-frequency signal and delay is mixed
Conjunction obtains mixed frequency signal, and handles the mixed frequency signal, so that each loudspeaker unit replay signal in loudspeaker array
Frequency response it is consistent.
Correspondingly, the playback module 2204 is specifically used for: by treated, mixed frequency signal is respectively sent to corresponding raise
In sound device unit, and the bass signal after delay is sent to bass effect unit, so that the bass effect unit resets institute
Bass signal after stating delay.
On the basis of the above embodiments, include the first audio signal and the second audio signal in received audio signal,
And when the first corresponding with the second audio signal Main beam angle difference of audio signal, then the playback module 2304 is specifically used
In: it will be mixed according to the first mixed frequency signal after first Audio Signal Processing and according to second after the second Audio Signal Processing
Frequency Signal averaging is simultaneously respectively sent in corresponding loudspeaker unit.
The device of audio processing provided in an embodiment of the present invention is suitable for the audio processing that above-mentioned any embodiment provides
Method has corresponding function and beneficial effect.
Note that the above is only a better embodiment of the present invention and the applied technical principle.It will be appreciated by those skilled in the art that
The invention is not limited to the specific embodiments described herein, be able to carry out for a person skilled in the art it is various it is apparent variation,
It readjusts and substitutes without departing from protection scope of the present invention.Therefore, although being carried out by above embodiments to the present invention
It is described in further detail, but the present invention is not limited to the above embodiments only, without departing from the inventive concept, also
It may include more other equivalent embodiments, and the scope of the invention is determined by the scope of the appended claims.
Claims (10)
1. a kind of method of audio processing characterized by comprising
It is followed by receiving audio signal opening projection, the audio signal is divided by high-frequency signal according to preset division point and low frequency is believed
Number；
It determines delay parameter, and is delayed according to the delay parameter to the low frequency signal, wherein the delay parameter packet
Include the determination at a distance from reflecting surface by mobile terminal；
By the high-frequency signal and delay after low frequency signal be mixed to get mixed frequency signal, and to the mixed frequency signal at
Reason, so that the frequency response of each loudspeaker unit replay signal is consistent in loudspeaker array；
By treated, mixed frequency signal is respectively sent in corresponding loudspeaker unit, so that after loudspeaker unit reproduction process
Mixed frequency signal.
2. the method according to claim 1, wherein the determining delay parameter includes:
Reflective distance parameter is obtained, and determines the delay parameter of the low frequency signal according to the reflective distance parameter, wherein institute
Stating reflective distance parameter is loudspeaker array at a distance from reflecting surface.
3. according to the method described in claim 2, it is characterized in that, described determine the low frequency according to the reflective distance parameter
The delay parameter of signal includes:
According to formula:Determine the delay parameter of the low frequency signal, wherein PFFor the reflection of acquisition
Distance parameter, Δ t are preset time parameter, tdFor delay parameter.
4. the method according to claim 1, wherein it is described to the mixed frequency signal carry out processing include:
Branch is carried out to the mixed frequency signal, wherein the number of branch is identical as the number of loudspeaker unit in loudspeaker array；
Every road mixed frequency signal is handled respectively.
5. according to the method described in claim 4, it is characterized in that, it is described respectively to every road mixed frequency signal carry out processing include:
Determine the filter factor of the corresponding filter of each loudspeaker unit；
Every road mixed frequency signal is respectively sent in corresponding filter, so that each filter is carried out according to the filter factor of itself
Filtering.
6. according to the method described in claim 5, it is characterized in that, each loudspeaker unit of the determination corresponding filter
Filter factor includes:
It obtains the frequency response of each loudspeaker unit, the target frequency response of loudspeaker array and loudspeaker array and resets sound
The Main beam angle of frequency signal；
According to each frequency response, target frequency response and the Main beam angle, LS solution of the least norm is utilized
Determine the frequency domain filtering coefficient of each filter；
It is corresponding time domain filter coefficients as the filter of each filter using each frequency domain filtering transformation of coefficient using inversefouriertransform
Wave system number.
7. according to the method described in claim 6, it is characterized in that, the formula of the LS solution of the least norm are as follows:
minimum(‖C·X-B‖2)=0, wherein Wherein, loudspeaker array includes N number of loudspeaker unit, Ω altogethermFor loudspeaker array
M-th of acoustic beam angle of playback audio signal, m=(1,2 ..., tar ..., M) wherein, ΩtarFor the loudspeaker battle array of acquisition
The Main beam angle of column playback audio signal, remaining is the audio signal angle of radiation in addition to Main beam, HnFor n-th of loudspeaking
The frequency response of device unit, n=(1,2 ..., N),For transmission of n-th of loudspeaker unit under m-th of acoustic beam angle
Function, XnFor the frequency domain filtering coefficient of the corresponding filter of n-th of loudspeaker unit, H0For the target of the loudspeaker array of acquisition
Frequency response, B (Ωm) it is ΩmThe acoustic pressure of upper audio signal, k are the discrete value of the audible frequency of the loudspeaker array of setting.
8. the method according to claim 1, wherein described be divided into the audio signal according to preset division point
After high-frequency signal and low frequency signal further include:
The low frequency signal is divided into low frequency subsignal and bass signal according to default bass part-frequency point；
Correspondingly, the low frequency signal by after the high-frequency signal and delay is mixed to get mixed frequency signal and includes:
Low frequency subsignal after the high-frequency signal and delay is mixed to get mixed frequency signal；
Correspondingly, described will treated that mixed frequency signal is respectively sent in corresponding loudspeaker unit includes:
By treated, mixed frequency signal is respectively sent in corresponding loudspeaker unit, and the bass signal after delay is sent to
Bass effect unit, so that the bass effect unit resets the bass signal after the delay.
9. the method according to claim 1, wherein including the first audio signal and the in received audio signal
Two audio signals, and when corresponding with the second audio signal Main beam angle difference of the first audio signal, it is described will treated
Mixed frequency signal is respectively sent in corresponding loudspeaker unit
After according to the first mixed frequency signal after first Audio Signal Processing and according to second Audio Signal Processing
Second mixed frequency signal is superimposed and is respectively sent in corresponding loudspeaker unit.
10. a kind of device of audio processing characterized by comprising
The audio signal is divided into height according to preset division point for being followed by receiving audio signal in unlatching projection by frequency division module
Frequency signal and low frequency signal；
Time delay module is delayed to the low frequency signal for determining delay parameter, and according to the delay parameter, wherein
The delay parameter includes being determined at a distance from reflecting surface by mobile terminal；
Processing module, for the low frequency signal after the high-frequency signal and delay to be mixed to get mixed frequency signal, and to described mixed
Frequency signal is handled, so that the frequency response of each loudspeaker unit replay signal is consistent in loudspeaker array；
Playback module, for will treated that mixed frequency signal is respectively sent in corresponding loudspeaker unit so that loudspeaker list
Mixed frequency signal after first reproduction process.
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