CN102760437A - Audio decoding device of control conversion of real-time audio track - Google Patents
Audio decoding device of control conversion of real-time audio track Download PDFInfo
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Abstract
The invention discloses an audio decoding device with configurable control conversion of a real-time audio track, in the technical field of digital audio processing. The audio decoding device comprises an audio track control unit, an address control unit, a logical calculus unit and a coefficient storage unit, wherein the audio track control unit is respectively connected with the coefficient storage unit and the address control unit and respectively used for outputting configuration parameter information and original audio information, the coefficient storage unit and the address storage unit are respectively connected with the logical calculus unit and used for outputting logic parameters and data addresses, and the logical calculus unit receives PCM data outputted by a decoder and outputs PCM sample values. The audio decoding device of control conversion of the real-time audio track realizes direct control and conversion of a decoding result, and obtains an anticipant output. The audio decoding device has the advantages of simple logic module, convenience for realizing address block operation in parallel, rapid conversion, synchronization and true time with a system on chip, and no delay.
Description
Technical field
What the present invention relates to is a kind of device of digital audio processing technical field, specifically is a kind of audio decoding apparatus with configurable real-time sound channel control transformation.
Background technology
Natural sound because the orientation of the difference of simulating signal sound source, sound generating is different, different apart from the distance of people's ear, characterize with the audio file of different sound channel standards later on being transformed into DAB so.At present, the form of DAB is formed with monophony, stereophony and surround sound.Monophonic audio is meant that the back is unified to be noted it by recording equipment mixing from the sound signal of different azimuth, is reset by an audio amplifier again.Dual-channel audio promptly is with two equipment of raising one's voice at an angle to each other; Each equipment of raising one's voice provides signal by a sound channel separately; The Biological Principles of the sound that the signal mode apery ear in the sound channel is heard at nature; Come the sound of simulating nature circle neutral body, be actually with 2D signal and come the three-dimensional source of sound in the simulating reality world.Surround sound is a kind of multichannel audio.Normally compare, refer to sound like a kind of manner of playback that is surrounded the hearer with stereophony.The playback sound field that this mode produced except the Sounnd source direction sense that is keeping original signal, is also supervened around the acoustics of sense and expansion sense (sound leaves hearer's diffusion or the sensation of reverberation is arranged).
In many application backgrounds, require in certain distortion range, to change or to carry out the control of sound channel output between voice data by above various sound channel standards, with the following various examples that are applied as:
The amount of equipment of raising one's voice and mass deficiency need be mixed the multichannel audio that decoding draws and reduced to two-channel or monophonic audio, utilize the existing equipment of raising one's voice to carry out the output of audio frequency.
In the song audio frequency, hope only output accompaniment or the part of singing opera arias, the i.e. a certain channel information of an output audio.
After the decoding to the monophony of certain low tonequality or dual-channel audio, hope to export the audio frequency of a three-dimensional surround sound through computing.
In the audio frequency aftertreatment, multitrack audio file decoding back to be hoped the waveform of certain independent sound channel is modified, the data linear combination that perhaps utilizes certain several sound channel is to reach the effect of audio mixing.
Based on the conversion between the sound signal of above various sound channel standards, based on following several kinds of operations:
1, multiplication multiply by a constant value to certain channel data, promptly in the process of linear combination, gives weight to this channel data;
2, addition superposes the data of certain several sound channel;
3, displacement, the gain that causes in the reduction step in front;
4, saturated, prevent that the result from overflowing.
Below provide the example that 5.1 sound channel surround sound audio conversion are dual-channel audio: suppose existing 5.1 sound channel surround sound audio files, storage mode is a linear PCM.
Existing hope converts thereof into two-channel linear PCM data.
Suppose that the L channel data are left in 5.1 sound channels; The intermediate channel data are center; Right data is right, and a left side is left_sur around data, and right is right_sur around data; The L channel data are left_sample in the dual-channel audio of output, and right data is right_sample.Change through following method.
left_sample＝Saturate{[Shift(left×A)+Shift(center×B)+Shift(left_sur×C)]}
right_sample＝Saturate{[Shift(right×A)+Shift(center×B)+Shift(right_sur×C)]}
A, B, C are multiplication coefficient in this formula, are the constant that prestores, and the Shift ecbatic is by location number displacement, and the Saturate ecbatic carries out operated in saturation by certain bit wide.Obtain by the converted stereophony audio frequency of 5.1 sound channel surround sounds with this method.
In common real-time decoding Play System, the channel information of decoded output signal is identical with the side information of encoding compression file, even and if have the sound channel modular converter also to depend on the software code in the system, underaction and configurability are poor.The present invention provides the output channels control module when satisfying general decoder architecture, output channels is controlled; And because its configurability; Utilize hardware directly to realize various conversion easily, and need not write the code of software control, and because it directly utilizes decoded results to carry out synchronous arithmetic logical operation and output; As far as SOC(system on a chip), can accomplish real time output data.
Retrieval through to prior art is found; One Chinese patent application numbers 200580022064.0; Open day on June 6th, 2007, put down in writing a kind of " stereo set and reproduction mode setting method ", this technology comprises the processing and control element (PCE) 11 control demoders 21 of microcomputer 1; With the decoding input signal, and obtain the voice format information of input signal.Said processing and control element (PCE) 11 is based on obtaining the disposal route that voice format information is invoked at the matrix processing of storage in the storage unit 12.Said processing and control element (PCE) 11 obtains the coding channel information from input signal further.When said coding sound channel was two-channel, said processing and control element (PCE) 11 control loops were handled input signal is carried out matrix around processor 22, and output multichannel voice.
But this prior art can only be to the audio coding standard of some kinds of forms that configure; And depend on audio mode and decide reproducing mode; Relatively poor in dirigibility, be unfavorable for that the expansion after new audio standard occurs is used, and the related sound channel mapping algorithm of this disposal route has only been contained matrix operation; And comprised increasing nonlinear algorithm in the present audio algorithm, therefore the reproduction audio frequency of quality can not more be arranged.The most important thing is that the audio track transfer algorithm in this disposal route only only is made up of the default algorithm of its storage, can not be disposed by the user, therefore, dirigibility has great restriction.To this, this paper has proposed a kind of new sound channel converting means.
Summary of the invention
The present invention is directed to the above-mentioned deficiency that prior art exists, a kind of audio decoding apparatus of real-time sound channel control transformation is provided, can directly control and conversion the output that obtains expecting decoded result.Logic module is simple, and block operations Parallel Implementation is easily pressed in the address, realizes conversion apace, accomplish with SOC(system on a chip) synchronously with can produce delay in real time and not.
The present invention realizes through following technical scheme; The present invention includes: sound channel control module, address control unit, ALU and coefficient storage unit; Wherein: the sound channel control module links to each other with address control unit with the coefficient storage unit respectively and exports configuration parameter information and original audio information respectively; The coefficient storage unit links to each other with ALU respectively and output logic parameter and data address with address control unit, and the PCM data of ALU Rcv decoder output are also exported the PCM sample value.
Described sound channel control module comprises: sound channel control register and data message register; Wherein: built-in original channel sum of sound channel control register and channel properties thereof and output linear PCM sound channel sum and channel properties thereof; The bit wide of the built-in PCM data of data message register and the bit wide of PCM sample value; The sound channel control register is connected with address control unit, ALU, coefficient storage unit and transmits input and output channel information and about the configuration information of sound channel conversion, the data message register is connected with address control unit and transmitting audio data bit wide information.
Described sound channel control register internal memory contains: the related data that whether adopts decision bit, input and the output channels of default configuration algorithm; This related data comprises: the number of sound channel and the specific object of sound channel; Be respectively L channel, intermediate channel, R channel etc. in order; Corresponding bit is put the data of 1 this sound channel of expression and is participated in computing, is that 0 expression does not use the data of this sound channel and unnecessary bit to keep.
Described data message register memory contains: the data bit width of the data bit width of input PCM data and output PCM data.
Described address control unit comprises: two address base registers and two group address generation units; Wherein: the output of the first address base register carrying recorded decoding process comprises the address, memory block of original channel data message; First scalar/vector corresponds respectively to the data address of each separate channels of input; The memory buffer of second address base register record output PCM; Terminal device is taken data away from this buffer zone; What second scalar/vector pointed to is that the conversion sound channel of choosing in the sound channel control register afterwards enables corresponding sound channel address; A same PCM bit wide at interval between each sound channel initial value; The first address base register links to each other with ALU and transmits the PCM data storage addresses plot information that decoded result produces; The second address base register links to each other with ALU and transmits the PCM data storage addresses plot information after the sound channel conversion, and first scalar/vector links to each other with sound channel control register, ALU and transmits the address information of the current input of frame PCM data, and second scalar/vector links to each other with sound channel control register, ALU and transmits the address information of the current PCM of output of frame data.
Described ALU obtains the data address that contains input data address and output data address from first scalar/vector and second scalar/vector respectively; And from sound channel control register acquisition logic parameter; This ALU is made up of multiplier, totalizer, shift unit and saturation arithmetic unit, and wherein: multiplier, shift unit and the totalizer, the saturation unit that are complementary with number of channels are linked in sequence.
Described coefficient storage unit is built-in: multiplier enables, the multiplication coefficient of each sound channel, shift unit enable, the displacement coefficient of each sound channel, totalizer enable, saturation unit enables, the saturated width of each operation result.
The course of work of the present invention is following:
The first step, to different audio coding forms, decode and export corresponding each sound channel PCM data sample by demoder.The user is configured by form the sound channel control register, input sound channel sum and the channel type expected, and output channels sum and channel type, and select to adopt the configuration of acquiescence coefficient or the scheme of User Defined placement algorithm.
Second step, address control unit obtain PCM sample value plot from demoder, obtain PCM data bit width and sound channel sum and which sound channel participation logical operation from the sound channel control module, are begun to calculate each data address of a frame by the scalar/vector on both sides.
The 3rd step, logical block are obtained the input data source address and the data export target address of once-through operation from address control unit.The coefficient storage unit if take the acquiescence mode, then extracts default algorithm from sound channel control module acquisition algorithm configuration information from storage unit, will enable to dispose to logical block with coefficient, and the data of appropriate address are carried out exporting after the computing.As what take is the mode of User Defined configuration, then needs user's mode one group of vector of pre-stored in the coefficient storage unit by regulation, drives logical block with this vector, and the data of appropriate address are carried out exporting after the computing.
The present invention has the following advantages:
1. the present invention places real-time decoding system, directly decoded result is controlled and conversion the output that obtains expecting.Logic module is simple, and block operations Parallel Implementation is easily pressed in the address, realizes conversion apace, accomplish with SOC(system on a chip) synchronously with can produce delay in real time and not.
2. adopt the mode that register is set to come administration module, abundant and user interactions, and on output and mapping algorithm, realize configurable.
3. have very high extensibility, only need spreading coefficient memory module and sound channel control register, and the just following more processing of multichannel audio of completion of quantity that increases scalar/vector.And, support following emerging mapping algorithm through revising the structure of some hardware cell.
4. support multiple sound channel output neatly, both merely exported the data of certain several sound channel, the linear combination of also exporting certain several sound channel is also accomplished the algorithm of certain function and is realized the sound channel conversion.Only need revise register value and not relate to the control on the change software, be easy to realize.
5. this device does not relate to the demoder result of apparatus for processing audio front end, and only the result to demoder output carries out conversion, has wide range of applications, and makes IP easily.
Description of drawings
Fig. 1 is the synoptic diagram of apparatus for processing audio.
Fig. 2 is a sound channel control register synoptic diagram.
Fig. 3 is a scalar/vector calculated address process synoptic diagram.
Fig. 4 is the operational process synoptic diagram of ALU and arithmetic element.
Embodiment
Elaborate in the face of embodiments of the invention down, present embodiment provided detailed embodiment and concrete operating process, but protection scope of the present invention is not limited to following embodiment being to implement under the prerequisite with technical scheme of the present invention.
Embodiment
As shown in Figure 3; The audio decoding apparatus of the real-time sound channel control transformation of present embodiment comprises: sound channel control module 1, address control unit 2, ALU 3 and coefficient storage unit 4; Wherein: sound channel control module 1 links to each other with address control unit 2 with coefficient storage unit 4 respectively and exports configuration parameter information and original audio information respectively; Coefficient storage unit 4 links to each other with ALU 3 respectively and output logic parameter and data address with address control unit 2, and the PCM data of ALU 3 Rcv decoders 5 outputs are also exported the PCM sample value.
Described sound channel control module 1 comprises: sound channel control register 5 and data message register 6; Wherein: sound channel control register 5 built-in original channel sums and channel properties and output linear PCM sound channel sum and channel properties thereof; The bit wide of data message register 6 built-in PCM data and the bit wide of PCM sample value; Sound channel control register 5 is connected with address control unit 2, ALU 3, coefficient storage unit 4 and transmits input and output channel information and about the configuration information of sound channel conversion, data message register 6 is connected with address control unit 2 and transmitting audio data bit wide information.
Described sound channel control register 5 internal memories contain: the related data that whether adopts decision bit, input and the output channels of default configuration algorithm; This related data comprises: the number of sound channel and the specific object of sound channel; Be respectively L channel, intermediate channel, R channel etc. in order; Corresponding bit is put the data of 1 this sound channel of expression and is participated in computing, is that 0 expression does not use the data of this sound channel and unnecessary bit to keep.
Described data message register 6 internal memories contain: the data bit width of the data bit width of input PCM data and output PCM data.
Described address control unit 2 comprises: two address base registers 7,8 and two group address generation units 9,10; Wherein: the output of the first address base register, 7 carrying recorded decoding processes comprises the address, memory block of original channel data message; First scalar/vector 9 corresponds respectively to the data address of each separate channels of input; The memory buffer of the second address base register, 8 record output PCM; Terminal device is taken data away from this buffer zone; What second scalar/vector 10 pointed to is that the conversion sound channel of choosing in the sound channel control register 5 afterwards enables corresponding sound channel address; A same PCM bit wide at interval between each sound channel initial value; The first address base register 7 links to each other with ALU 3 and transmits the PCM data storage addresses plot information that decoded result produces; The second address base register 8 links to each other with ALU 3 and transmits the PCM data storage addresses plot information after the sound channel conversion, and first scalar/vector 9 links to each other with sound channel control register 5, ALU 3 and transmits the address information of the current input of frame PCM data, and second scalar/vector 10 links to each other with sound channel control register 5, ALU 3 and transmits the address information of the current PCM of output of frame data.
Described ALU 3 obtains the data address that contains input data address and output data address from first scalar/vector 9 and second scalar/vector 10 respectively; And from sound channel control register 5 acquisition logic parameters; This ALU 3 is made up of multiplier, totalizer, shift unit and saturation arithmetic unit, and wherein: multiplier, shift unit and the totalizer, the saturation unit that are complementary with number of channels are linked in sequence.
Described coefficient storage unit 4 is built-in: multiplier enables, the multiplication coefficient of each sound channel, shift unit enable, the displacement coefficient of each sound channel, totalizer enable, saturation unit enables, the saturated width of each operation result.
When sound channel control register 5 is set up and after data message register 612 obtained the bit wide of PCM data, scalar/vector 8 generates the address that will participate in the PCM data that sound channel changes according to above information.Sound channel before the conversion of choosing in the sound channel control register 5 will enable the first corresponding scalar/vector 9; The initial value of address enable unit each other be spaced apart 1 PCM bit wide; By the situation series arrangement that enables to choose, sum equals the input sound channel sum in the sound channel control register 5.The address of each scalar/vector 8 will be updated after each data output computing, and method for updating is forward direction sequential addressing input sound channel sum * PCM data bit width unit;
When the value of first scalar/vector 9 of participating in computing has been crossed the address boundary of storage area, and real-time decoding still continuing, and then returns the position of its initial value, repeats this process.The memory buffer of the second address base register, 8 record output PCM, terminal device is taken data away from this buffer zone.What second scalar/vector 10 pointed to is that the conversion sound channel of choosing in the sound channel control register 5 afterwards enables corresponding sound channel address; Same 1 PCM bit wide at interval between each sound channel initial value; Output channels number after last one group of Data Update in the forward direction sequential addressing sound channel control register a 5 * PCM bit wide unit, when first scalar/vector, 9 recorded address run off the border of output buffer, and decoding is still continuing; Just return its initial position, repeat this process again.The process of first scalar/vector, 9 calculated addresses is with reference to figure 3.
Described ALU 3 obtains the input data address from first scalar/vector 9, obtains the output data address from second scalar/vector 10, and controls which signal by sound channel control register 5 and participate in transform operation.
The required logic parameter of described coefficient storage unit 4 stored logic arithmetic elements 3; Those parameters will be participated in computing and determined by the input/output signal number that sound channel control register 5 is write down; Whether from sound channel control register 5, obtain information configured simultaneously; Deposit coefficient that coefficient replace original algorithm by specified format at first by the user, realize the function of configurable conversion coefficient, concrete method is:
Form a vector according to the number of input sound channel and output channels, control the dimension of input and output, for the required parameter of each ALU 3, like the multiplication constant of multiplier, the displacement coefficient of shift unit, the saturated bit wide of saturation unit etc.The form of storage unit is according to the format of first cell enable, unit, back coefficient.Coefficient storage unit 4 is stored enabling and coefficient vector of one group of acquiescence respectively by various input and output dimension; Do not carry out the adaptive of logical operation by this acquiescence vector under the situation of configuration voluntarily the user; When the user hopes by oneself algorithm output to be provided, deposit some groups of vectors in by this form.
Produced stereophony as the music of CD disc is decoded, only hoped now to export left channel signals in real time, promptly accompaniment music is provided with so as follows.Obviously, the input and output dimension is 2,1, needs to use multiplier; Multiplier enable with coefficient be 1,1,0, then the output of multiplier is exactly L channel; Shift unit, totalizer and saturation unit all do not enable, and are set to 0, x, x; X is indifferent numerical value, then pass through logical operation after, demoder output directly becomes the monophonic audio that the accompaniment L channel is only arranged by stereophony.
The real-time decoding device need carry out real-time decoding to the 5.1 sound channel surround sounds of AC3; Because the playback equipment hardware resource is not enough; Only support the broadcast of stereo audio; For obtaining the real-time play of optimum efficiency tonequality, directly adopt AC3down mix algorithm that the multichannel data are mixed to the linear PCM of demoder output and reduce to two-channel, export the stereo audio of each frame in play end in real time.
At first, by format configuration sound channel control register 5 shown in Figure 2, need to use the algorithm that oneself disposes; This Bit Allocation in Discrete is 1, and the input sound channel number is 6, and what need the participation computing is 5 sound channels; L channel, intermediate channel, R channel, a left side around, right around; The flag bit assignment of these 5 sound channels is 1, and the subwoofer sound channel is 0, and the input sound channel number is 3 ' b101.The output channels number is 2, is respectively L channel and R channel, and then its corresponding bit is 1, and all the other sound channel corresponding bit are 0, and the input sound channel number is 2 ' b10.The instruction that utilizes audio process to this register according to above explanation assignment.
Demoder has got into the intermediate storage zone to the resulting 5.1 sound channel PCM data of AC3 audio decoder; Suppose that this regional plot is 0x10000; According to the structure of demoder, data message register 6 is noted the bit wide of PCM data, is assumed to be 64 bits; The hypothesis decoding device once only transmits a frame to decoded result, and then the size in intermediate storage zone is set to 0x1C000.5.1 the linear PCM of sound channel is stored as sequential storage; The address base register 7 of input end is set to 0x10000 so; There are 5 scalar/vectors 8 to begin to calculate the address date of each computing; Might as well suppose the address of scalar/vector 81 record L channel data, the address of scalar/vector 82 record intermediate channel data, the address of scalar/vector 83 record right data; The address of the left surround channel data of scalar/vector 84 records, the address of the right surround channel data of scalar/vector 85 records.The initial value of scalar/vector 81 is 0x10000 so; The initial value of scalar/vector 82 is the PCM bit wide that 0x10000 adds 1 unit; Promptly be 0x10008; The initial value of scalar/vector 83 is 0x10010, and the initial value of scalar/vector 84 is 0x10018, and the initial value of scalar/vector 85 is 0x10020.The address base of output terminal is assumed to be 0x20000; The storage format of two-channel linear PCM is a sequential storage; Have two scalar/vectors 8 to begin to calculate the address date of each computing, what might as well establish scalar/vector 86 calculating is the address of output L channel data, and the initial value of scalar/vector 86 is 0x20000 so; What scalar/vector 87 write down is the address of output right data, and the initial value of scalar/vector 87 is 0x20002 so.
By the down mix algorithm of AC3 by the following pseudo-code that provides:
left_out＝Saturate{Shift_Scale{Shift_Twid[(left)＊DOWNMIX_FACTOR]+Shift_Twid[(center)＊LUTS_PARAM]+Shift_Twid[(left_sur)＊LUTC_PARAM]}}
right_out＝Saturate{Shift_Scale{Shift_Twid[(right)＊DOWNMIX_FACTOR]+Shift_Twid[(center)＊LUTS_PARAM+Shift_Twid(right_sur)＊LUTC_PARAM]}}
Left_out, right_out are for exporting the L channel data and the right data of two-channel in the example, and left, center, right, left_sur, right_sur import L channel data, intermediate channel data, right data, left surround channel data and the right surround channel data of 5.1 sound channels respectively.DOWNMIX_FACTOR, LUTS_PARAM, LUTC_PARAM are respectively and take advantage of property coefficient; Provide by coefficient storage unit 4; Shift_Sclae and Shift_Twid are respectively according to Scale width and Twid width and are shifted, and this displacement width equally can be by 4 storages of coefficient storage unit.Saturate is the operated in saturation to the result, and saturated width is by 4 storages of coefficient storage unit.
Two groups of initial vectors of existing storage are given coefficient storage unit 4, and primary vector is for calculating the value of output L channel, and it is 1 that multiplier enables; Coefficient is respectively the L channel coefficient, and DOWNMIX_FACTOR is 0x3504 in the present embodiment; Intermediate channel coefficient LUTS_PARAM is 0x257a in the present embodiment, and the R channel coefficient is 0; Left side surround channel coefficient is LUTC_PARAM, is 0x1a82 in the present embodiment, and right surround channel coefficient is 0.It is 1 that shift unit enables, and each sound channel displacement coefficient is Twid, is 15 in the present embodiment.It is 1 that totalizer enables, and the addition results displacement is 15 in the present embodiment for Scale, and it is 1 that saturation unit enables, and saturated width is output PCM sample value width 16.Second group of vector is for calculating the value of output R channel, and it is 1 that multiplier enables, and it is 0 that coefficient is respectively the L channel coefficient, intermediate channel coefficient LUTS_PARAM; Be 0x257a in the present embodiment, the R channel coefficient is DOWNMIX_FACTOR, is 0x3504 in the present embodiment, and doing the surround channel coefficient is 0; Right surround channel coefficient is LUTC_PARAM, is 0x1a82 in the present embodiment, and it is 1 that shift unit enables, and each sound channel displacement coefficient is Twid; Be 15 in the present embodiment, it is 1 that totalizer enables, and the addition results displacement is Scale; Be 15 in the present embodiment, it is 1 that saturation unit enables, and saturated width is output PCM sample value 16.
The scalar/vector 8 of input end continues to calculate next addresses; According to rule, next L channel address, promptly scalar/vector 81 last institute recorded address add that the input sound channel sum in the sound channel control register 5 multiply by input PCM bit wide; Be 0x10000+6 * 0x8; Be 0x100, the recording address of scalar/vector 82 is 0x10038, and the recording address of scalar/vector 83 is 0x100; The recording address of scalar/vector 84 is 0x10048, and the recording address of scalar/vector 85 is 0x10050.Output terminal scalar/vector 8 also begins to calculate the address of next group output; The OPADD of location generation unit 86 multiply by output PCM bit wide for last address, this unit adds the output channels sum in the sound channel control register 5 by rule; Be 0x20000+2 * 0x2; Be 0x20004, the OPADD of scalar/vector 87 is 0x20006.Carry out next group computing by the address logic arithmetic element that refreshes 3, produce second group of PCM sample value.
Accomplish when the content of a frame, the calculated address of scalar/vector 8 will exceed the border, memory block, will reach 0x1c000 like the address of the scalar/vector 81 of record L channel data address; Just reach between the limit; Mean ED, will accept the next frame of demoder this moment and accomplish signal, return relevant recorded address for the first time with plot; Be 0x10000, the class of operation of remaining scalar/vector 8 seemingly.
So just with demoder synchronous real-time ground output dual-channel audio to the terminal equipment of raising one's voice.With 5.1 sound channel AC3 audio frequency through decoding after, directly produce stereophony to the terminal device plays of raising one's voice.
Through technological improvement of the present invention; After having adopted sound channel control module 1 of the present invention; Can the sound channel of any dimension be carried out conversion according to user configured method, conveniently management, and the mode of conversion and feasibility are not restricted by audio coding form and feasibility; Because it is form, also irrelevant with upper application software by hardware cell.Simultaneously, logical block provided by the invention has contained matrix operation and nonlinear operation, and better more perfect support can be provided mapping algorithm now, makes that the sound quality after the conversion is better.
Claims (7)
1. the audio decoding apparatus of a real-time sound channel control transformation; It is characterized in that; Comprise: sound channel control module, address control unit, ALU and coefficient storage unit; Wherein: the sound channel control module links to each other with address control unit with the coefficient storage unit respectively and exports configuration parameter information and original audio information respectively; The coefficient storage unit links to each other with ALU respectively and output logic parameter and data address with address control unit, and the PCM data of ALU Rcv decoder output are also exported the PCM sample value.
2. the audio decoding apparatus of real-time sound channel control transformation according to claim 1; It is characterized in that; Described sound channel control module comprises: sound channel control register and data message register; Wherein: built-in original channel sum of sound channel control register and channel properties thereof and output linear PCM sound channel sum and channel properties thereof; The bit wide of the built-in PCM data of data message register and the bit wide of PCM sample value; The sound channel control register is connected with address control unit, ALU, coefficient storage unit and transmits input and output channel information and about the configuration information of sound channel conversion, the data message register is connected with address control unit and transmitting audio data bit wide information.
3. the audio decoding apparatus of real-time sound channel control transformation according to claim 1 is characterized in that, described sound channel control register internal memory contains: the related data that whether adopts decision bit, input and the output channels of default configuration algorithm; This related data comprises: the number of sound channel and the specific object of sound channel; Be respectively L channel, intermediate channel, R channel etc. in order; Corresponding bit is put the data of 1 this sound channel of expression and is participated in computing, is that 0 expression does not use the data of this sound channel and unnecessary bit to keep.
4. the audio decoding apparatus of real-time sound channel control transformation according to claim 1 is characterized in that, described data message register memory contains: the data bit width of the data bit width of input PCM data and output PCM data.
5. the audio decoding apparatus of real-time sound channel control transformation according to claim 1; It is characterized in that; Described address control unit comprises: two address base registers and two group address generation units; Wherein: the output of the first address base register carrying recorded decoding process comprises the address, memory block of original channel data message; First scalar/vector corresponds respectively to the data address of each separate channels of input; The memory buffer of second address base register record output PCM; Terminal device is taken data away from this buffer zone, and what second scalar/vector pointed to is that the conversion sound channel of choosing in the sound channel control register afterwards enables corresponding sound channel address, a same PCM bit wide at interval between each sound channel initial value; The first address base register links to each other with ALU and transmits the PCM data storage addresses plot information that decoded result produces; The second address base register links to each other with ALU and transmits the PCM data storage addresses plot information after the sound channel conversion, and first scalar/vector links to each other with sound channel control register, ALU and transmits the address information of the current input of frame PCM data, and second scalar/vector links to each other with sound channel control register, ALU and transmits the address information of the current PCM of output of frame data.
6. the audio decoding apparatus of real-time sound channel control transformation according to claim 1; It is characterized in that; Described ALU obtains the data address that contains input data address and output data address from first scalar/vector and second scalar/vector respectively; And from sound channel control register acquisition logic parameter; This ALU is made up of multiplier, totalizer, shift unit and saturation arithmetic unit, and wherein: multiplier, shift unit and the totalizer, the saturation unit that are complementary with number of channels are linked in sequence.
7. the audio decoding apparatus of real-time sound channel control transformation according to claim 1; It is characterized in that described coefficient storage unit is built-in: multiplier enables, the multiplication coefficient of each sound channel, shift unit enable, the displacement coefficient of each sound channel, totalizer enable, saturation unit enables, the saturated width of each operation result.
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| CN112788350A (en) * | 2019-11-01 | 2021-05-11 | 上海哔哩哔哩科技有限公司 | Live broadcast control method, device and system |
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| CN106663434A (en) * | 2014-06-27 | 2017-05-10 | 杜比国际公司 | Method for determining for the compression of an hoa data frame representation a lowest integer number of bits required for representing non-differential gain values |
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| CN111656442A (en) * | 2017-11-17 | 2020-09-11 | 弗劳恩霍夫应用研究促进协会 | Apparatus and method for encoding or decoding directional audio coding parameters using quantization and entropy coding |
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| CN111756940A (en) * | 2020-07-07 | 2020-10-09 | 广州威谱通信设备有限公司 | Simplified digital voice communication system with programmable addressing and double-input sound mixing |
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