CN106231489A - The treating method and apparatus of audio frequency - Google Patents
The treating method and apparatus of audio frequency Download PDFInfo
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- CN106231489A CN106231489A CN201610589851.3A CN201610589851A CN106231489A CN 106231489 A CN106231489 A CN 106231489A CN 201610589851 A CN201610589851 A CN 201610589851A CN 106231489 A CN106231489 A CN 106231489A
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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
- H04R1/00—Details of transducers, loudspeakers or microphones
- H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
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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
- H04R5/00—Stereophonic arrangements
- H04R5/04—Circuit arrangements, e.g. for selective connection of amplifier inputs/outputs to loudspeakers, for loudspeaker detection, or for adaptation of settings to personal preferences or hearing impairments
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Abstract
The invention discloses the treating method and apparatus of a kind of audio frequency.Wherein, the method includes: obtain the sampling number of each sound channel in multiple sound channel；According to sampling number, the audio signal of each sound channel is sampled, obtain sampled data；Obtain filter data；It is filtered processing to sampled data according to filter data.The present invention solves the technical problem that when playing audio frequency by earphone in prior art, some position acoustics is different.
Description
Technical field
The present invention relates to audio area, in particular to the treating method and apparatus of a kind of audio frequency.
Background technology
On market, the film audio of main flow is made as 5.1 sound channels or 7.1 sound channels, is all the speaker by different orientations
Realize playback.
5.1 sound channels refer to center channel, preposition left and right sound channels, rearmounted left and right cincture sound channel, and so-called 0.1 sound channel weight
Subwoofer channel.Set of system can connect 6 loudspeaker altogether.Center channel loudspeaker, are responsible for regeneration and coordinate the action on screen, big portion
Between timesharing, it is responsible for the part of personage's dialogue；Preposition left and right sound channels loudspeaker, then be used to make up beyond center Screen or not
The action can seen from screen and other sound；Rearmounted cincture sound effect loudspeaker is responsible for peripheral and whole background music, allows people feel
The centre of whole scene, the effect that the shock such as surged ahead, aircraft roar past are placed oneself in the midst of in feel from the crown；And motor
The supper bass of shake people's heartstrings such as sound, the sound of bomber or big drum, then completed by mega bass loudspeaker.This cover system excellent
Point is to obtain the vivid and broader sound field of sound apparent before, fabulous sound field and real solid surround,
Such that it is able to the trickle sound listened in unprecedented background moves.
A set of sound field relative equilibrium front and back is set up in the effect of 7.1 sound channel systems in simple terms around hearer
Sound field, is different from 5.1 sound channel sound fields, it add on the basis of original after in sound field sound channel, simultaneously it also different from
Common 6.1 sound channel sound fields because 7.1 sound channels have two-way after in put, and after this two-way in the maximum effect put be contemplated to prevent
Hearer is acoustically producing the deviation of sound field because not being sitting in emperor position.
In different positions, acoustics is different, and the acoustics in some positions (such as emperor position) is preferable, and
Acoustics in other position (such as, movie theatre or the position at arenas edge) is poor.
For above-mentioned problem, effective solution is the most not yet proposed.
Summary of the invention
Embodiments provide the treating method and apparatus of a kind of audio frequency, at least to solve prior art is passed through ear
The technical problem that during machine sowing playback frequency, some position acoustics is different.
An aspect according to embodiments of the present invention, it is provided that the processing method of a kind of audio frequency, including: obtain multiple sound channel
In the sampling number of each sound channel；According to described sampling number, the audio signal of described each sound channel is sampled, adopted
Sample data；Obtain filter data；It is filtered processing to described sampled data according to described filter data.
Further, obtain filter parameter to include: load wave filter file；The information comprising file header solves
Analysis；Read the data in described wave filter file according to the information parsed, obtain described filter data.
Further, described wave filter file is multiple.
Further, before loading wave filter file, described method also includes: determine that needs add according to the quantity of sound channel
The quantity of the wave filter file carried.
Further, obtain the sampling number of each sound channel in multiple sound channel to include: according to the significant figure of wave filter file
According to total bytes, the quantity of sound channel that described wave filter file comprises, each sampled point needs bit number calculate each sound
The sampling number in road.
Further, the sound channel comprised according to the total bytes of valid data of wave filter file, described wave filter file
Quantity, the bit number that needs of each sampled point calculates the sampling number of each sound channel and includes: be calculated according to the following equation every
The sampling number of individual sound channel: the total bytes ÷ of the valid data of sampling number=8 of each sound channel × wave filter file is (described
The bit number of the quantity of the sound channel that wave filter file comprises × each sampled point needs).
Further, after obtaining sampled data, described method also includes: described sampled data is carried out FFT,
Obtain the audio sampling data of frequency domain；According to described filter data, described sampled data is filtered process to include: at frequency
In territory, described filter data is multiplied with described audio sampling data；The data obtained being multiplied afterwards carry out FFT inverse transformation,
Wherein, the data carrying out obtaining after FFT inverse transformation are plural numbers；Data treating excess syndrome portion that FFT inverse transformation obtains will be carried out and export.
Another aspect according to embodiments of the present invention, additionally provides the processing means of a kind of audio frequency, including: first obtains list
Unit, for obtaining the sampling number of each sound channel in multiple sound channel；Sampling unit, for according to described sampling number to described often
The audio signal of individual sound channel is sampled, and obtains sampled data；Second acquisition unit, is used for obtaining filter data；Filtering is single
Unit, for being filtered processing to described sampled data according to described filter data.
Further, described second acquisition unit includes: add subelements, is used for loading wave filter file；Resolve son single
Unit, resolves for the information comprising file header；Read subelement, for reading described filtering according to the information parsed
Data in device file, obtain described filter data.
Further, described wave filter file is multiple.
Further, described device also comprises determining that unit, for described add subelements load wave filter file it
Before, the quantity of the wave filter file needing loading is determined according to the quantity of sound channel.
Further, described first acquiring unit includes: obtain subelement, for the valid data according to wave filter file
Total bytes, the quantity of sound channel that described wave filter file comprises, each sampled point needs bit number calculate each sound channel
Sampling number.
Further, described acquisition subelement includes: computing module, for each sound channel is calculated according to the following equation
Sampling number: total bytes ÷ (the described wave filter literary composition of the valid data of sampling number=8 of each sound channel × wave filter file
The bit number of the quantity of the sound channel that part comprises × each sampled point needs).
Further, described device also includes: converter unit, is used for after described sampling unit obtains sampled data,
Described sampled data is carried out FFT, obtains the audio sampling data of frequency domain；Described filter unit includes: first calculates son
Unit, for being multiplied described filter data with described audio sampling data in frequency domain；Second computation subunit, being used for will
The data obtained afterwards that are multiplied carry out FFT inverse transformation, and wherein, the data carrying out obtaining after FFT inverse transformation are plural numbers；Output
Unit, for will carry out data treating excess syndrome portion that FFT inverse transformation obtains and export.
In embodiments of the present invention, in multiple sound channels, each sound channel has several sampling numbers, the sound to each sound channel
Frequently signal is sampled, and obtains sampled data, by different filter data simulation optional position sound sources, according to wave filter number
According to sampled data is filtered process, after having processed, no matter the voice data obtained in which position is play, and effect is all
It is identical, the technique effect that when having reached to play audio frequency, all positions acoustics is identical, and then solve in prior art
The technical problem that when playing audio frequency by earphone, some position acoustics is different.
Accompanying drawing explanation
Accompanying drawing described herein is used for providing a further understanding of the present invention, constitutes the part of the present invention, this
Bright schematic description and description is used for explaining the present invention, is not intended that inappropriate limitation of the present invention.In the accompanying drawings:
Fig. 1 is the flow chart of the processing method of audio frequency according to embodiments of the present invention；
Fig. 2 is the general flow chart of system according to embodiments of the present invention；
Fig. 3 is the execution flow chart of function initPLL_SDRAM () according to embodiments of the present invention；
Fig. 4 is the execution flow chart of function InitDAI () according to embodiments of the present invention；
Fig. 5 is the topology diagram of I2S signal according to embodiments of the present invention；
Fig. 6 is the execution flow chart of function enable_SPORT () according to embodiments of the present invention；
Fig. 7 is the execution flow chart of decoder module according to embodiments of the present invention；
Fig. 8 is the schematic diagram of the call relation between primary function according to embodiments of the present invention；
Fig. 9 is the process chart of data processing module according to embodiments of the present invention；
Figure 10 is the principle schematic of overlap-add method according to embodiments of the present invention；
Figure 11 be overlap-add method according to embodiments of the present invention implement schematic diagram；
Figure 12 is the schematic diagram of the processing means of audio frequency according to embodiments of the present invention.
Detailed description of the invention
In order to make those skilled in the art be more fully understood that the present invention program, below in conjunction with in the embodiment of the present invention
Accompanying drawing, is clearly and completely described the technical scheme in the embodiment of the present invention, it is clear that described embodiment is only
The embodiment of a present invention part rather than whole embodiments.Based on the embodiment in the present invention, ordinary skill people
The every other embodiment that member is obtained under not making creative work premise, all should belong to the model of present invention protection
Enclose.
It should be noted that term " first " in description and claims of this specification and above-mentioned accompanying drawing, "
Two " it is etc. for distinguishing similar object, without being used for describing specific order or precedence.Should be appreciated that so use
Data can exchange in the appropriate case, in order to embodiments of the invention described herein can with except here diagram or
Order beyond those described is implemented.Additionally, term " includes " and " having " and their any deformation, it is intended that cover
Cover non-exclusive comprising, such as, contain series of steps or the process of unit, method, system, product or equipment are not necessarily limited to
Those steps clearly listed or unit, but can include the most clearly listing or for these processes, method, product
Or intrinsic other step of equipment or unit.
First the technical term involved by the embodiment of the present invention is made description below:
WAV form: WAV is a kind of AIFC of Microsoft exploitation, and it meets RIFF (Resource
Interchange File Format) filespec, for preserving the audio-frequency information resource of windows platform, by Windows
Platform and application program thereof are extensively supported, this form also supports the multiple compaction algorithms methods such as MSADPCM, CCITT ALAW,
Holding multiple digital audio, sampling frequency and sound channel, the wav file of standard format is the same with CD form, is also the sampling of 44.1K
Frequency, 16 quantify numeral, therefore very nearly the same in audio files quality and CD！WAV is closest to lossless music format, institute
The biggest with file size.The advantage of WAV audio format includes: simple coding/decoding (almost directly storage from
The signal of A/D converter (ADC)), universal approval/support and lossless storage.The major defect of WAV form is needs
Audio storage space.For little storage restriction or little bandwidth applications, this is probably an important problem.WAV form
Another one latent defect be that 2G in 32 wav files limits, this restriction is at the W64 developed for SoundForge
Form is improved.
Hrir and hrtf:HRTF (Head Related Transfer Function): head related transfer function, is a kind of
Audio location algorithm, the HRIR (Head Related Inpulse Response) of corresponding time domain, head related impulse response.People
Having two ears, but can position from three-dimensional sound, this benefits from the human ear analysis system to acoustical signal.HRTF can
To be decomposed into three parts, Interaural Time Difference, Interaural Level Difference and
Spectral Cues.The signal passing to human ear (before tympanum) from space any point can describe with a filtering system,
Acoustical signal before what source of sound+wave filter obtained is exactly two ear drum membranes.This transmission system is a flight data recorder, and we need not close
Heartfelt wishes sound is how to be delivered to ears, and only need to be concerned about the difference of source of sound and binaural signal.Describe if we obtain this group
The wave filter (transmission function) of spatial information, i.e. HRTF, just can reduce from the acoustical signal in this orientation, space (as double in passed through
Sound channel earphone).If we have all orientation, the space bank of filters to ears, just can obtain a filtering matrix, thus go back
Originally from the acoustical signal of whole dimensional orientation.But a severe problem is that HRTF is highly personal, at laboratory
The group of functions recording everyone is unpractical.Therefore, HRTF technology is directed to study and the most truly reduces individual
The spatial function changed.This is also the target that virtual reality (virtual reality technology) wishes to reach.
I2S:I2S (Inter IC Sound) bus, also known as integrated circuit built-in audio bus, is that PHILIPS Co. is
A kind of bus standard that voice data between digital audio-frequency apparatus transmits and formulates, this bus is specialized between audio frequency apparatus
Data are transmitted, and are widely used in various multimedia system.It has employed setting along independent wire transmission clock and data signal
Meter, by separating data with clock signal, it is to avoid because of the distortion of time difference induction, saves purchase for user and resists audio frequency and tremble
The expense of dynamic professional equipment.I2S has 3 main signals: 1. serial clock SCLK, is also bit clock (BCLK), i.e. corresponding number
Each data of word tone frequency, SCLK has 1 pulse.Frequency=2 of SCLK × sample frequency × sampling resolution.2. frame clock
LRCK, (also referred to as WS), for the data of switching right and left sound channel.For " 1 ", LRCK represents that transmitting is the data of R channel, for
" 0 " then represents that transmitting is the data of L channel.The frequency of LRCK is equal to sample frequency.3. serial data SDATA, it is simply that
The voice data represented by the complement of two's two's complement.Sometimes for making can preferably synchronize between system, in addition it is also necessary to additionally transmission one
Signal MCLK, referred to as master clock, be also system clock (Sys Clock), is 256 times or 384 times of sample frequency.Along with technology
Development, under unified I2S interface, occur in that multiple different data form.According to SDATA data relative to LRCK and
The position of SCLK is different, is divided into left-justify (less use), I2S form (form that i.e. Philip specifies) and Right Aligns (also to cry
Japan form, common format).
According to embodiments of the present invention, it is provided that the embodiment of the processing method of a kind of audio frequency, it should be noted that at accompanying drawing
The step shown in flow chart can perform in the computer system of such as one group of computer executable instructions, and, although
Show logical order in flow charts, but in some cases, can with the order being different from herein perform shown or
The step described.
Fig. 1 is the flow chart of the processing method of audio frequency according to embodiments of the present invention, as it is shown in figure 1, the method include as
Lower step:
Step S102, obtains the sampling number of each sound channel in multiple sound channel.
Step S104, samples to the audio signal of each sound channel according to sampling number, obtains sampled data.
Step S106, obtains filter data.
Step S108, is filtered processing to sampled data according to filter data.
In embodiments of the present invention, in multiple sound channels, each sound channel has several sampling numbers, the sound to each sound channel
Frequently signal is sampled, and obtains sampled data, by different filter data simulation optional position sound sources, according to wave filter number
According to sampled data is filtered process, after having processed, no matter the voice data obtained in which position is play, and effect is all
It is identical, solves the technical problem that when playing audio frequency by earphone in prior art, some position acoustics is different, reach
Arrive the technique effect that when playing audio frequency, all positions acoustics is identical.
Embodiments provide the process using two kinds of methods of DSP and FPGA that audio frequency is processed, by virtual
Multichannel optional position sound source, achieves the virtualization effect of multichannel surround sound with DSP or FPGA.This cover system excellent
Point is to obtain the vivid and broader sound field of sound apparent before, fabulous sound field and real solid surround,
Such that it is able to the trickle sound listened in unprecedented background moves.
Below the process using DSP method to process audio frequency is described in detail.
The embodiment of the present invention is described the functional module of DSP program in detail and is divided and the execution flow process of modules, change
Amount and primary function etc..
Running environment: system is made up of structure cabinet, board and DSP program code three part, and wherein board includes base plate
With core board two parts.DSP program code operates in the DSP on core board.
The main-process stream of system is as shown in Figure 2.As in figure 2 it is shown, this flow process comprises the following steps:
Step S202, DSP hardware initializes.After powering on, first system carries out the hardware initialization work of DSP.
Step S204, reads wave filter file and decodes.After initialization completes, all filtering can used below to be read
The data of device.The data of these wave filter presented in wav file, therefore will by these files read in DSP, and from
File solves real data, with the process of input signal later.
Step S206, reads input data.Audio signal is input in real time, and is that multichannel is simultaneously entered.
Step S208, processes input data.DSP carries out Filtering Processing after reading part input data.
Step S210, the data after output process.The signal processed is sent in DAC input, under reading the most again
One group of input data.This process performs continuously, at the signal that outfan is just processed and synthesizes.
Can be 4 modules by whole procedure division as described above: hardware initialization module, wav decoder module,
Data processing module and input/output module.Below these four modules are described in detail.
1, hardware initialization module.
Initialization module includes the initialization of PLL and SDRAM, DAI initialization, initialization three part of SPORT.
1.1, data describe
(1) variable pPMCTL
PPMCTL is pointer variable, points to depositor PMCTL, is used for configuring the parameter of PLL.
(2) variable pSYSCTL
PSYSCTL is pointer variable, points to depositor SYSCTL, is used for enabling memory and selects.
(3) variable pEPCTL
PEPCTL is pointer variable, points to depositor EPCTL, for Mapping B ank0 to SDRAM.
(4) variable pSDCTL
PSDCTL is pointer variable, points to depositor SDCTL, is used for configuring the running parameter of SDRAM.
(5) variable pSDRRC
PSDRRC is pointer variable, points to depositor SDRRC, the configuration in terms of arranging refresh rate and reading optimization.
(6) variable SPORTx_FS_I, SPORTx_CLK_I and SPORTx_DA_I
When SPORTx is configured to I2S pattern, these three signal represents the frame synchronization of I2S bus, clock and data respectively.
(7) variables D AI_PBx_I and DAI_PBx_O
The two signal represents the input of DAI pin buffer respectively and pin buffer exports.Make during external signal input
With DAI_PBx_O, during internal signal output, use DAI_PBx_I.
(8) variable pSPMCTLx
PSPMCTLx is pointer variable, points to depositor SPMCTLx, for multi-way contral.
(9) variable pSPCTLx
PSPCTLx is pointer variable, points to depositor SPCTLx, for SPORT control.
(10) variable pDIVx
PDIVx is pointer variable, points to depositor DIVx, for arranging the frequency dividing ratio of clock and frame synchronizing signal.
(11) variable pCPSPxA or pCPSPxB
PCPSPxA or pCPSPxB is pointer variable, points to depositor CPSPxA or CPSPxB.CPSPxA or CPSPxB refers to
DMA parameter is organized to next.
(12) array receivetcb [RX_SPORT_NUM] [RX_I2S_NUM] [4]
Receivetcb defines and receives each I2S data wire all correspondences one of TCB, each reception SPORT individually
TCB。
(13) array transmttcb [TX_SPORT_NUM] [TX_I2S_NUM] [4]
Transmttcb defines and sends each I2S data wire all correspondences one of TCB, each transmission SPORT individually
TCB。
1.2, function describes
(1) function initPLL_SDRAM ()
The function of this function is to initialize PLL and SDRAM respectively.Initialization procedure sets gradually PMCTL,
Five depositors of SYSCTL, EPCTL, SDCTL and SDRRC.The function of these depositors is all described in previous joint.This function
Execution flow process as shown in Figure 3.
As it is shown on figure 3, the execution flow process of this function comprises the following steps:
Step S302, arranges the locking frequency of PLL.
Step S304, enables memory and selects.
Step S306, allows Bank0 be mapped to SDRAM.
Step S308, arranges SDRAM running parameter.
Step S310, arranges refresh rate, reads to optimize.
Step S302 is used for carrying out PLL initialization.Step S304 to step S310 is used for carrying out SDRAM initialization.
(2) function InitDAI ()
The function of this function is to initialize DAI, is exactly specifically to distribute DAI pin for signal.First by institute after powering on
Have the input of pin buffer and pin buffer to enable input to drag down.Then input pattern is selected.The design has two kinds defeated
Entering module available: one is SPDIF pattern, another kind is AES/EBU pattern, and select here is latter mode.By
In be up to 8 tunnel input in the design, it is therefore desirable to 4 receive two-way input for the SPORT, each SPORT received.With
Time output only have two-way (left and right acoustic channels), need 1 SPORT to be used for sending.It is respectively these SPORT distribution interface signals.DSP
Data after process are sent to DAC by SPORT, DAC distribution interface signal to be.ADC Yu DSP in the design
The interface of interface and DSP Yu DAC all uses I2S.For ADC, DSP works in from pattern；For DSP, DAC
Also work in from pattern.The execution flow process of this function is as shown in Figure 4.
As shown in Figure 4, this flow process comprises the following steps:
Step S402, DAI pin status initializes.
Step S404, input pattern selects.
Step S406, SPORT0 signal distributes.
Step S408, SPORT1 signal distributes.
Step S410, SPORT2 signal distributes.
Step S412, SPORT3 signal distributes.
Step S414, SPORT4 signal distributes.
Step S416, DAC signal distributes.
SPORT1 to SPORT4 is 4 SPORT for reception.By performing step S402 to step S416, it is possible to just
Beginningization DAI.
In the design, the chip for the decoding of AES/EBU signal is CS8416, and each CS8416 chip all produces one group
I2S signal.Due to multiple signals two-way the to be synthesized output of input, therefore select the clock in first group of I2S signal
The common control signal that signal and frame synchronizing signal work as each SPORT and DAC, the topological structure of I2S signal in system
As shown in Figure 5.
(3) function enable_SPORT ()
The function of this function is to initialize SPORT, completes both sides work: one is that depositor is arranged, and two is that TCB sets
Put.In TCB is arranged, specify the length of DMA transfer data and reception or send the address of relief area.Holding of this function
Row flow process is as shown in Figure 6.
As shown in Figure 6, the execution flow process of this function comprises the following steps:
Step S602, multi-way contral depositor resets.
Step S604, SPORT controls depositor and resets.
Step S606, SPORT frequency dividing depositor resets.
Step S608, receives SPORT TCB and arranges.
Step S610, sends SPORT TCB and arranges.
Step S612, SPORT controls depositor and arranges.
By execution step S602 to step S612, reach to initialize the purpose of SPORT.
2, input/output module detailed design
2.1, data describe
(1) grand RX_SPORT_NUM
RX_SPORT_NUM defines the maximum number receiving SPORT, due to system compatible 5.1 sound channel and 7.1 sound channels,
Many have 8 tunnel inputs simultaneously, it is therefore desirable to 4 receive SPORT.
(2) grand TX_SPORT_NUM
TX_SPORT_NUM defines the number sending SPORT, the filtered device of multipath audio signal of input process after
The output of synthesis two-way, delivers to left and right ear respectively eventually, and the SPORT therefore sent only needs one.
(3) grand SIZE_DMA_BLOCK
SIZE_DMA_BLOCK defines the size of dma operation of certain passage data block to be transmitted, namely one
The sampled point number that passage is once sampled.
(4) grand SIZE_DMA_BUFF
SIZE_DMA_BUFF defines the size receiving and sending DMA buffer, and reception and the transmission of DSP are all passed through
I2S completes, and every data line can transmit two passages, and therefore the size of DMA buffer is that DMA transfer data block is big
Little twice.
(5) array rx_SPORT_Buf [RX_SPORT_NUM] [SIZE_DMA_BUFF]
Rx_SPORT_Buf defines reception SPORT relief area.The data wire that receives of each reception SPORT all has correspondence
Relief area.
(6) array tx_SPORT_Buf [TX_SPORT_NUM] [SIZE_DMA_BUFF]
Tx_SPORT_Buf defines transmission SPORT relief area.The data wire that sends of each transmission SPORT all has correspondence
Relief area.
(7) array In_split_Buf [IN_CHAN_NUM] [SIZE_DMA_BLOCK]
In_split_Buf defines reception and splits relief area, and each input sound channel has the relief area of correspondence.
(8) array Out_unite_Buf [SIDE_OF_OUT] [SIZE_DMA_BLOCK]
Out_unite_Buf defines transmission and merges relief area, and left and right acoustic channels is respectively arranged with the relief area of correspondence.
(9) array In_Chan_Buf [IN_CHAN_NUM] [FFT_POINT_NUM]
In_Chan_Buf defines input data buffer zone, and data the most to be processed (including 0 data supplemented) store
In this relief area.
(10)
Array Out_Chan_Buf [IN_CHAN_NUM] [SIDE_OF_OUT] [SIZE_DMA_BLOCK]
Out_Chan_Buf defines data output buffer district, and the data after process of convolution are stored in this relief area.
(11) array Out_Bkup_Buf [IN_CHAN_NUM] [SIDE_OF_OUT] [SIZE_DMA_BLOCK]
Out_Bkup_Buf defines output backup buffers, and this relief area realizes the function of overlap-add for auxiliary.
2.2, function describes
(1) function Receive ()
Parameter: channel number
Return value: nothing
Function describes: the function of this function is from I2S data of a SPORT of input by the data of certain sound channel
Line is separated.Repeatedly call this function and can obtain 8 input sound channels from 4 data line of outside input.
(2) function Transmit ()
Parameter: nothing
Return value: nothing
Function describes: the function of this function is that the data sending two and merging relief area are merged into the one of transmission SPORT
On data line, thus exported by outside DAC.
3, decoder module detailed design
Decoder module is a more complicated module in the design.The execution flow process of this module is as shown in Figure 7.Such as figure
Shown in 7, the execution flow process of this module comprises the following steps:
Step S702, resolves and initializes.
Step S704, opens file.
Step S706, document analysis.
Step S708, reads one group of data.
Step S710, data type conversion.
Step S712, it may be judged whether run through all data.If it is judged that be yes, perform step S714；
If it is judged that be no, perform step S708.
Step S714, it may be judged whether run through all passages.If it is judged that be yes, perform step S716；If it is determined that
Result is no, performs step S708.
Step S716, closes closed file.
Step S718, it may be judged whether complete bank of filters and load.If it is judged that be yes, terminate；If it is judged that
It is no, performs step S702.
3.1, the form of wave filter file
3.1.1, the form of wav file
WAVE file is made up of several Chunk.Generally comprise according to appearance position hereof: RIFF
WAVE Chunk, Format Chunk, Fact Chunk (optional), Data Chunk.
The form of RIFF WAVE Chunk is as follows:
The form of Fact Chunk is as follows:
The form of Data Chunk is as follows:
The bit position of wav data is segmented into following several form:
3.1.2 wave filter file content example
Wav file can be opened with binary form.The H0e000a.wav that with binary system software for editing open is presented herein below
File.The information of file header can be parsed easily according to introduction above.Valid data according to wave filter file
The quantity of the sound channel that total bytes, wave filter file comprise, the bit number of each sampled point needs calculate the sampling of each sound channel
Count.Specifically, the sampling number of each sound channel it is calculated according to the following equation: sampling number=8 of each sound channel × filtering
Total bytes ÷ (the bit of the quantity of the sound channel that wave filter file comprises × each sampled point needs of the valid data of device file
Number).
For example, it is assumed that wave filter file comprises two sound channels, each sampled point 16bit, valid data 512 byte altogether
(noticing that this document uses little-Endian form).The sampling number that thus can calculate each sound channel is 128
Point.
3.2 data describe
(1) grand WAV_BUF_CHANS
WAV_BUF_CHANS defines the port number of wave filter file, knowable to analysis above, and these wave filter files
Being all double track, therefore value is 2.
(2) grand WAV_BUF_SAMPS
WAV_BUF_SAMPS defines the number of wave filter file each sound channel sampled point, specific number see above point
Analysis.
(3) array hrtf_data_buffer [MAX_FLT_NUM] [WAV_BUF_CHANS] [WAV_BUF_SAMPS]
Hrtf_data_buffer defines pool of buffer district, the filter data after depositing FFT, each filter
The corresponding single relief area of each sound channel of ripple device file.
(4) array hrir_data_buffer_ptrs [WAV_BUF_CHANS]
Hrir_data_buffer_ptrs defines an array of pointers, and each pointer in array points to a filtering
The data_buffe relief area that device file is corresponding.
(5) array hrir_data_buffer_strides [WAV_BUF_CHANS]
Hrir_data_buffer_strides defines an array, and the element in array represents and reading a filtering
Span size during data_buffer buffer data corresponding to device file, in the design, these value perseverances are 1.
3.3 functions describe
(1) function load_hrir_group ()
Parameter: nothing
Return: nothing
Function: load wave filter file according to the way of input sound channel and resolve, with resolving at the beginning of the filter data obtained
The corresponding filter buffer of beginningization.Wave filter file is multiple.Before loading wave filter file, the quantity according to sound channel is true
The quantity of the fixed wave filter file needing to load.If system inputs with 5.1 sound channels, this function can load and resolve 6 filtering
Device file, last wave filter file is used for processing subwoofer input；If system inputs with 7.1 sound channels, this function can add
Carry and resolve 8 wave filter files, equally last wave filter file to be used for processing subwoofer input.
(2) function load_hrir_file ()
Parameter: parameter 1, wave filter title；Parameter 2, filter data buffer storage.
Return: nothing
Function: according to incoming wave filter title, loads wave filter file and resolves, and data parsing obtained store
In the buffer storage specified.
(3) function wavFileReadOpen ()
Parameter: parameter 1, points to the pointer of wav document instance type；Parameter 2, filename.
Return: perform result return code.
Function: open and resolve the wav file specified.
(4) function wavFileReadClose ()
Parameter: point to the pointer of wav document instance type.
Return: perform result return code.
Function: the wav file of the front opening of closedown.
(5) function decode_hrir_file ()
Parameter: parameter 1, points to the pointer of wav document instance type；Parameter 2, the sampling number of each passage；Parameter 3,
Point to the pointer of data buffer zone.
Return: nothing
Function: read the data of specified file dedicated tunnel having already turned on, and these data are stored specify slow
Rush in district.
(6) function adi_wav_parse_init ()
Parameter: point to the pointer of wav file information structure body.
Return: perform result return code.
Function: initialize the file information structure body of a wav document instance.
(7) function parse_wav_file ()
Parameter: point to the pointer of wav document instance type.
Return: an integer, represent execution state.
Function: resolve a wav file having already turned on.
(8) function adi_wav_parser ()
Parameter: parameter 1, points to the pointer of wav file information structure body；Parameter 2, the byte number of input block；Parameter
3, point to the pointer of incoming bit stream；Parameter 4, inputs byte offset value.
Return: perform result return code.
Function: most basic wav document analysis subprogram.
(9) function wavFileRead ()
Parameter: parameter 1, points to the pointer of wav document instance type；Parameter 2, points to the second rank pointer of channel buffer；
Parameter 3, the span of channel buffer；Parameter 4, the sampling number of each passage.
Return: the number of the word read.
Function: read a channel data of wav file and be stored in data buffer zone and return reads the number of this channel word
Mesh.
3.4, function calling relationship
Call relation between primary function is as shown in Figure 8.The most each wave filter file is main after being loaded
Yao Youliang great call relation branch: the major function of left side branch is the parsing of information comprised to wav file header, medial fascicle
Function be the data that information according to file header reads in file, and carry out data type conversion as requested, filtered
Device data.
4, data processing module detailed design
The handling process of data processing module is as shown in Figure 9.
As it is shown in figure 9, this handling process comprises the following steps:
Step S902, calculates extended filtering device data 128FFT.The FFT first having to calculate all wave filter to be used becomes
Changing, this step completes initializing during wave filter, so can save in real time process below
Between timesharing.
Step S904, initializes complex data relief area.Entrance real time process first has to initialize complex data and delays
Rush district, prepare for complex operation below.
Step S906, calculates 128 FFT of input audio data.When the sampling number of each sound channel is 128, initialize
After completing, 128 audio sampling datas to input calculate FFT, obtain 128 point data of frequency domain.
Step S908, with filter data multiplication operation in frequency domain.By data and the audio sample of wave filter in frequency domain
Data are multiplied.
Step S910, the data after computing carry out 128 inverse FFT.Data after being multiplied remake FFT inverse transformation.
Step S912, the real part of data exporting after taking conversion.Data after FFT inverse transformation are plural number, take out real part also
Output.
From step S906 to step S912, after having initialized, sampled data is carried out FFT, obtain frequency domain
Audio sampling data；According to filter data, sampled data is filtered process to include: by filter data in frequency domain
It is multiplied with audio sampling data；The data obtained being multiplied afterwards carry out FFT inverse transformation, wherein, obtain after carrying out FFT inverse transformation
To data be plural number；Data treating excess syndrome portion that FFT inverse transformation obtains will be carried out and export.
4.1, data describe
(1) grand IN_LINE_NUM
IN_LINE_NUM defines the number receiving data wire.For 5.1 sound channels, there are 6 input sound channels, IN_
LINE_NUM is 3；For 7.1 sound channels, having 8 input sound channels, IN_LINE_NUM is 4.
(2) grand CH_PER_LINE
CH_PER_LINE defines the number of channels that every input data line is carried, due to connecing between DSP and ADC
Mouth is I2S, and therefore every data line can carry two sound channels.
(3) grand IN_CHAN_NUM
IN_CHAN_NUM defines the number of input sound channel, and this number is straight with IN_LINE_NUM and CH_PER_LINE
Connect relevant.
(4) grand MAX_FLT_NUM
MAX_FLT_NUM defines the number of hrir wave filter, in the ordinary course of things this number and the channel number of input
Mesh keeps consistent, it is also possible to inconsistent, depending on this will be according to concrete application scenario.Note each wave filter mentioned here
Including left and right ear.
(5) grand SIDE_OF_OUT
SIDE_OF_OUT defines the channel number that output data line is carried, and exports only two sound channels in left and right in the design,
Therefore value is 2.
(6) array hrtf_buf
Hrtf_buf defines wave filter input data buffer zone, is the data in frequency domain, complex data type.
(7) array audio_buf
Audio_buf defines audio input data relief area, is the data in time domain, floating point type.
(8) array fft_buf
Fft_buf defines audio frequency conversion data buffer zone, is the data in frequency domain, complex data type.
(9) array conv_buf
Conv_buf defines the time domain data relief area after process, floating point type.
4.2, function describes
(1) function multi_input ()
Parameter: nothing.
Return: nothing.
Function: realize filtering and the complex functionality of the process of multiple input path, i.e. multiple input path.This function is whole journey
The process core of sequence.
(2) function conv_with_hrir ()
Parameter: parameter 1, filter data, is the data after FFT；Parameter 2, audio sampling data, after being sampling
Initial data；Parameter 3, is filtered output data, is time domain data.
Return: nothing.
Function: realize the Filtering Processing of one group of input audio data, i.e. convolution algorithm.
Problems is typically had to two kinds of convolution algorithms: overlap-add method and overlap-save method, both algorithm function
Quite, sampling former algorithm in the design.The principle of this algorithm is as shown in Figure 10.
In algoritic module, overlap-add method implements as shown in figure 11.As shown in figure 11, the voice data of input
Sampling 64 points, gather into 128 point data by zero padding every time.Decoded filter data, the data of each passage are 128 points.
Although the partial data of wave filter is 128 points, but the most front 64 point data included wave filter all effectively
Information.Here 128 point data of wave filter are intercepted front 64 point data, still gathered into the data of 128 by zero padding.Audio frequency
128 point data and wave filter 128 point data carry out FFT respectively, and the data after conversion are multiplied in a frequency domain, and product is carried out
The inverse transformation of FFT, obtain or the data of 128.This 128 point data is divided into first half and latter half.First 64
Data are added with the latter half of last operation result (for first time computing it is believed that the result of last computing is
Zero), the result as this computing exports；The data of latter 64 are stored temporarily in relief area for next execution cycle.
This process moves in circles, so that infinite.
Embodiments provide the process using two kinds of methods of DSP and FPGA that audio frequency is processed, above section
The process using DSP method to process audio frequency has been described in detail.Use two kinds of methods of FPGA to audio frequency at
The process of reason and the similar process using DSP method to process audio frequency, repeat no more.
According to embodiments of the present invention, the processing means of a kind of audio frequency is additionally provided.The processing means of this audio frequency can perform
The processing method of above-mentioned audio frequency, the processing method of above-mentioned audio frequency can also be implemented by the processing means of this audio frequency.
Figure 12 is the schematic diagram of the processing means of audio frequency according to embodiments of the present invention.As shown in figure 12, this device includes
First acquiring unit 10, sampling unit 20, second acquisition unit 30 and filter unit 40.
First acquiring unit 10, for obtaining the sampling number of each sound channel in multiple sound channel.
Sampling unit 20, for sampling the audio signal of each sound channel according to sampling number, obtains sampled data.
Second acquisition unit 30, is used for obtaining filter data.
Filter unit 40, for being filtered processing to sampled data according to filter data.
Alternatively, second acquisition unit 30 includes adding subelements, resolving subelement and read subelement.Add carrier list
Unit, is used for loading wave filter file.Resolve subelement, resolve for the information that file header is comprised.Read subelement, use
In reading the data in wave filter file according to the information parsed, obtain filter data.
Alternatively, wave filter file is multiple.
Alternatively, device also includes determining unit.Determine unit, for add subelements load wave filter file it
Before, the quantity of the wave filter file needing loading is determined according to the quantity of sound channel.
Alternatively, the first acquiring unit 10 includes obtaining subelement.Obtain subelement, for having according to wave filter file
The effect total bytes of data, the quantity of sound channel that wave filter file comprises, the bit number of each sampled point needs calculate each sound
The sampling number in road.
Alternatively, obtain subelement and include computing module.Computing module, is used for each sound channel is calculated according to the following equation
Sampling number: total bytes ÷ (the wave filter file of the valid data of sampling number=8 of each sound channel × wave filter file
The bit number of the quantity of the sound channel comprised × each sampled point needs).
Alternatively, device also includes converter unit.Converter unit, is used for after sampling unit 20 obtains sampled data,
Sampled data is carried out FFT, obtains the audio sampling data of frequency domain.Filter unit 40 include the first computation subunit,
Two computation subunit and output subelement.First computation subunit, is used for filter data and audio sample number in frequency domain
According to being multiplied.Second computation subunit, carries out FFT inverse transformation for the data obtained being multiplied afterwards, wherein, carries out FFT inversion
The data obtained the most afterwards are plural numbers.Output subelement, for will carry out data treating excess syndrome portion that FFT inverse transformation obtains and export.
In the above embodiment of the present invention, the description to each embodiment all emphasizes particularly on different fields, and does not has in certain embodiment
The part described in detail, may refer to the associated description of other embodiments.
In several embodiments provided by the present invention, it should be understood that disclosed technology contents, can be passed through other
Mode realizes.Wherein, device embodiment described above is only schematically, the division of the most described unit, Ke Yiwei
A kind of logic function divides, actual can have when realizing other dividing mode, the most multiple unit or assembly can in conjunction with or
Person is desirably integrated into another system, or some features can be ignored, or does not performs.Another point, shown or discussed is mutual
Between coupling direct-coupling or communication connection can be the INDIRECT COUPLING by some interfaces, unit or module or communication link
Connect, can be being electrical or other form.
The described unit illustrated as separating component can be or may not be physically separate, shows as unit
The parts shown can be or may not be physical location, i.e. may be located at a place, or can also be distributed to multiple
On unit.Some or all of unit therein can be selected according to the actual needs to realize the purpose of the present embodiment scheme.
It addition, each functional unit in each embodiment of the present invention can be integrated in a processing unit, it is also possible to
It is that unit is individually physically present, it is also possible to two or more unit are integrated in a unit.Above-mentioned integrated list
Unit both can realize to use the form of hardware, it would however also be possible to employ the form of SFU software functional unit realizes.
If described integrated unit realizes and as independent production marketing or use using the form of SFU software functional unit
Time, can be stored in a computer read/write memory medium.Based on such understanding, technical scheme is substantially
The part that in other words prior art contributed or this technical scheme completely or partially can be with the form of software product
Embodying, this computer software product is stored in a storage medium, including some instructions with so that a computer
Equipment (can be for personal computer, server or the network equipment etc.) perform the whole of method described in each embodiment of the present invention or
Part steps.And aforesaid storage medium includes: USB flash disk, read only memory (ROM, Read-Only Memory), random access memory are deposited
Reservoir (RAM, Random Access Memory), portable hard drive, magnetic disc or CD etc. are various can store program code
Medium.
The above is only the preferred embodiment of the present invention, it is noted that for the ordinary skill people of the art
For Yuan, under the premise without departing from the principles of the invention, it is also possible to make some improvements and modifications, these improvements and modifications also should
It is considered as protection scope of the present invention.
Claims (14)
1. the processing method of an audio frequency, it is characterised in that including:
Obtain the sampling number of each sound channel in multiple sound channel；
According to described sampling number, the audio signal of described each sound channel is sampled, obtain sampled data；
Obtain filter data；
It is filtered processing to described sampled data according to described filter data.
Method the most according to claim 1, it is characterised in that obtain filter parameter and include:
Load wave filter file；
The information comprising file header resolves；
Read the data in described wave filter file according to the information parsed, obtain described filter data.
Method the most according to claim 2, it is characterised in that described wave filter file is multiple.
Method the most according to claim 3, it is characterised in that before loading wave filter file, described method also includes:
Quantity according to sound channel determines the quantity of the wave filter file needing loading.
Method the most according to claim 1, it is characterised in that obtain the sampling number bag of each sound channel in multiple sound channel
Include:
The quantity of the sound channel that the total bytes of the valid data according to wave filter file, described wave filter file comprise, each adopt
The bit number that sampling point needs calculates the sampling number of each sound channel.
Method the most according to claim 5, it is characterised in that according to the total bytes of the valid data of wave filter file,
The quantity of the sound channel that described wave filter file comprises, the bit number of each sampled point needs calculate the sampling number bag of each sound channel
Include:
The sampling number of each sound channel be calculated according to the following equation:
Total bytes ÷ (the described wave filter file bag of the valid data of sampling number=8 of each sound channel × wave filter file
The bit number of the quantity of the sound channel contained × each sampled point needs).
Method the most according to claim 6, it is characterised in that
After obtaining sampled data, described method also includes: described sampled data is carried out FFT, obtains the sound of frequency domain
Frequency sampling data；
According to described filter data, described sampled data is filtered process to include:
In frequency domain, described filter data is multiplied with described audio sampling data；
The data obtained being multiplied afterwards carry out FFT inverse transformation, and wherein, the data carrying out obtaining after FFT inverse transformation are plural numbers；
Data treating excess syndrome portion that FFT inverse transformation obtains will be carried out and export.
8. the processing means of an audio frequency, it is characterised in that including:
First acquiring unit, for obtaining the sampling number of each sound channel in multiple sound channel；
Sampling unit, for sampling the audio signal of described each sound channel according to described sampling number, obtains hits
According to；
Second acquisition unit, is used for obtaining filter data；
Filter unit, for being filtered processing to described sampled data according to described filter data.
Device the most according to claim 8, it is characterised in that described second acquisition unit includes:
Add subelements, be used for loading wave filter file；
Resolve subelement, resolve for the information that file header is comprised；
Read subelement, for reading the data in described wave filter file according to the information parsed, obtain described wave filter
Data.
Device the most according to claim 9, it is characterised in that described wave filter file is multiple.
11. devices according to claim 10, it is characterised in that described device also includes:
Determine unit, for loading before wave filter file at the described subelements that adds, determine that needs add according to the quantity of sound channel
The quantity of the wave filter file carried.
12. devices according to claim 8, it is characterised in that described first acquiring unit includes:
Obtain subelement, for the sound comprised according to the total bytes of valid data of wave filter file, described wave filter file
The bit number that the quantity in road, each sampled point need calculates the sampling number of each sound channel.
13. devices according to claim 12, it is characterised in that described acquisition subelement includes:
Computing module, for the sampling number of each sound channel is calculated according to the following equation:
Total bytes ÷ (the described wave filter file bag of the valid data of sampling number=8 of each sound channel × wave filter file
The bit number of the quantity of the sound channel contained × each sampled point needs).
14. devices according to claim 13, it is characterised in that
Described device also includes: converter unit, for after described sampling unit obtains sampled data, to described sampled data
Carry out FFT, obtain the audio sampling data of frequency domain；
Described filter unit includes:
First computation subunit, for being multiplied described filter data with described audio sampling data in frequency domain；
Second computation subunit, carries out FFT inverse transformation for the data obtained being multiplied afterwards, wherein, carries out FFT inversion alternatively
After the data that obtain be plural number；
Output subelement, for will carry out data treating excess syndrome portion that FFT inverse transformation obtains and export.
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