CN103107863B - Digital audio source coding method and device with segmented average code rate - Google Patents
Digital audio source coding method and device with segmented average code rate Download PDFInfo
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Abstract
The invention relates to a digital audio source coding method and device with a segmented average code rate. The method comprises the following steps: s1, determining the frame number N of the audio coding frame contained in a transmission frame according to the channel coding and modulation parameter information of the digital audio broadcasting system; s2, determining the total block number of the channel coding block contained in a transmission frame according to the channel coding and modulation parameter information of the digital audio broadcasting system; s3, determining the byte number of the source coding data in a channel coding block according to the channel coding and modulation parameter information of the digital audio broadcasting system; s4, calculating the total byte number of N audio coding frames contained in one transmission frame; s5, compressing and coding N frames of audio data continuously input in a transmission frame time, and adaptively allocating the byte number of each audio coding frame among the N audio coding frames based on the total byte number calculated in the step S4. The present invention can obtain a locally optimal compression within one transmission frame, thereby improving coding efficiency.
Description
Technical field
The present invention relates to the digital audio source coding technique of digital audio broadcast system, more particularly, the present invention relates to a kind of segmental averaging code check (SegmentedAverageBitRate proposed considering the feature of chnnel coding and modulation technique in digital audio broadcast system, and the digital audio source coding method of layer and section average bit rate (Layered & SegmentedAverageBitRate, LS-ABR) and device S-ABR).
Background technology
In the application systems such as current digital audio broadcasting, adopt the digital modulation technique of channel (as OFDM etc.) to require modulation signal has longer symbol lengths, therefore a modulation-frame will comprise multiple audio coding frame to form superframe structure; And general channel coding technology all adopts LDPC(LowDensityParityCheckCode in system, low density parity check code) Linear block coding, the data assembling of message sink coding becomes multiple integer information block (when adopting hierarchical coding, every layer of multiple integer information block of the pattern of wants) so that realize LDPC chnnel coding with regard to needing by this.
GB GB/T22726-2008 " multi-sound channel digital audio encoding and decoding technique specification ", be also called DRA(DigitalRiseAudio) audio standard, provide three kinds of coding modes, i.e. CBR(ConstantBitRate, normal bit rate), VBR(ViableBitRate, variable bit rate) and ABR(AverageBitRate, mean bit rate).Wherein, CBR pattern refers to that each coded frame has equally than the length of byte number; VBR pattern refers to that each coded frame can have any different byte length (certainly generally actual can limit the upper limit, in order to ensure that coding quality preferably also arranges lower limit when realizing); Abr mode make full use of coding side one have suitable size buffer (easily make coding delay increase too greatly, and decoding end buffer storage increase; Too littlely be unfavorable for that encryption algorithm is to encoded content smoothing processing), when ensureing that this buffer does not produce overflow and underflow, every frame length all alterable that coding exports, and population mean to get off be constant close to set by one, but for abr mode, if during with a certain integer frame combination, generally not there is same byte length.
International audio coding standard and equipment are also generally support above three kinds of coding modes, i.e. CBR, ABR and VBR, such as following common international code algorithm:
Layer I and the layer II of MPEG-1 and MPEG-2 only support CBR pattern;
The layer III of MPEG-1 and MPEG-2 supports VBR pattern, also can support CBR and abr mode;
DolbyAC-3 supports CBR pattern;
AAC and the MPEG surround sound of MPEG-2 and MPEG-4 supports CBR, VBR and abr mode.
At CMMB(ChinaMobileMultimediaBroadcasting, China Mobile multimedia broadcasting) in application, only can support the DRA audio coding of CBR pattern.Because every frame audio coding adopts fixed byte (or bit) length, in CMMB, chnnel coding adopts LDPC block encoding, require that the source coded data inputing to LDPC should be the data block of multiple regular length, therefore the DRA audio coding of CBR pattern makes whole system simplicity of design; But because audio signal content is complicated and changeable, therefore each signal frame (be 21ms to the DRA frame length of 48kHz audio frequency) scope adopts same information to represent, sound quality variation after easily causing every frame to encode, if under high code check, mass change is not easily discovered by people's ear, but for the application of CMMB low bit-rate, then there will be coding distortion when compressing complex audio signal, cause the problem occurring that subjective sound quality declines.
To the digital audio broadcasting (CDR that China is formulating, ChinaDigitalRadio) system standard, consider the importance of every frame information in the feature of its chnnel coding and digital modulation technique, digital broadcasting covering problem and message sink coding code stream and non-uniform Distribution etc., digital audio encoding preferably supports hierarchical coding, namely comprises core layer (or Primary layer) and enhancement layer; Or for non-layered audio coding, still a frame coded audio data can be divided into important (corresponding core layer) and insignificant data (for enhancement layer) two parts.Be convenient to chnnel coding like this and adopt non-technology such as error protection such as grade; namely high protection class is given to Primary layer; and give low protection class to enhancement layer; thus under complicated reception condition; can ensure that user correctly can receive the data of core layer (Primary layer); core layer audio-frequency unit can be recovered after decoding, ensure to listen to basic broadcast sounds quality.
But current digital audio encoder (or algorithm) generally only provides CBR, ABR and VBR Three models, when this Three models is applied to the CDR broadcast system in CMMB and future, there is following shortcoming:
1) if adopt CBR pattern, be limited to each audio frame and must be encoded to fixing byte length to represent the input audio signal of real-time dynamic change, overall message sink coding efficiency can be caused not high.
2) if directly encoder is set to ABR and VBR pattern, two kinds of situations can be divided into again:
I) when encoder is not stratified coding situation, due to the frame length (byte number) unfixing (change) of each audio coding frame, therefore in a transmission frame time, the total data of the multiple audio coding frames if desired transmitted is too much, then cannot sent by a transmission frame after Channel Coding and Modulation; Or the total data of the multiple audio coding frames if desired transmitted is very few, then make this transmission frame waste some code checks, and this can further improve the quality of the several audio frames in this transmission frame originally.
Ii) when encoder is layered coding case; the Primary layer of the several audio coding frames in a transmission frame and the respective total data of enhancement layer more uncontrollable; fluctuated; thus also can cause in certain transmission frame and cannot all to transmit and certain transmission frame there will be the situation of waste; especially to Primary layer and enhancement layer adopt do not wait error protection encode LDPC time, can become more complicated.
Summary of the invention
The technical problem to be solved in the present invention is, for the above-mentioned defect of prior art, provide a kind of digital audio source coding method and device of the segmental averaging code check for realizing forced coding efficiency in digital audio broadcast system when not increasing system complexity.
The technical solution adopted for the present invention to solve the technical problems is: the digital audio source coding method proposing a kind of segmental averaging code check, comprises the steps:
S1, determine the frame number N of the audio coding frame comprised in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
S2, determine the total block data of the channel coding blocks comprised in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
S3, determine the byte number of source coded data in a channel coding blocks according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
S4, calculate in a transmission frame total bytes of the N number of audio coding frame comprised based on step S2, S3;
S5, carry out compressed encoding to N number of audio coding frame of continuously input in the transmission frame time, the total bytes calculated based on step S4 distributes the byte number of each audio coding frame adaptively in N number of audio coding interframe.
In an embodiment, described step S1 determines the frame number of the audio coding frame comprised in a transmission frame in the following way:
N=T/t，
Wherein, N represents frame number; T represents the time span of a transmission frame of described digital audio broadcast system; T represents the time span of an audio coding frame of described digital audio broadcast system, is determined by the message sink coding mode of described digital audio broadcast system.
In an embodiment, described step S2 determines the total block data of the channel coding blocks comprised in a transmission frame in the following way:
K=A/b，
Wherein, K represents total block data; A represents the total bytes that a transmission frame of described digital audio broadcast system can transmit, by modulation system and the band bandwidth decision of described digital audio broadcast system; B represents the total bytes of a channel coding blocks, is determined by the channel coding method of described digital audio broadcast system.
In an embodiment, described step S4 calculates in a transmission frame total bytes of the N number of audio coding frame comprised in the following way:
B=K*M，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; K represents total block data; M represents the byte number of source coded data in a channel coding blocks.
In an embodiment, described method also comprises: determine whether to carry out hierarchical coding according to the modulation system of described digital audio broadcast system or channel coding method; And
When hierarchical coding, described step S2 comprises further:
Based on the respective channel coding method of Primary layer and enhancement layer and the channel coding blocks block number of modulation system determination Primary layer and the channel coding blocks block number of enhancement layer, and meet:
K=K
b+K
e，
Wherein, K represents total block data; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer;
Described step S4 calculates in a transmission frame total bytes of the N number of audio coding frame comprised in the following way:
B=B
b+B
e=K
b*M
b+K
e*M
e，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; B
brepresent the total bytes of the source coded data of Primary layer; B
erepresent the total bytes of the source coded data of enhancement layer; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer; M
brepresent the byte number of source coded data in the channel coding blocks of Primary layer; M
erepresent the byte number of source coded data in the channel coding blocks of enhancement layer.
In an embodiment, when hierarchical coding, described step S5 comprises further:
Compressed in layers coding is carried out to N number of audio coding frame of input continuously in the transmission frame time, the source coded data total bytes of the Primary layer calculated based on step S4 distributes the Primary layer byte number of each audio coding frame adaptively, and the source coded data total bytes of the enhancement layer calculated based on step S4 distributes the enhancement layer byte number of each audio coding frame adaptively.
The present invention solves the digital audio message sink coding device that its technical problem also proposes a kind of segmental averaging code check, comprising:
First computing module, for the frame number N of audio coding frame determining to comprise in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
Second computing module, for the total block data of channel coding blocks determining to comprise in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
3rd computing module, for determining the byte number of source coded data in a channel coding blocks according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
4th computing module, for calculating in a transmission frame total bytes of the N number of audio coding frame comprised based on the result of the second computing module and the 3rd computing module;
Message sink coding module, for carrying out compressed encoding to N number of audio coding frame of continuously input in the transmission frame time, the total bytes calculated based on the 4th computing module distributes the byte number of each audio coding frame adaptively in N number of audio coding interframe.
In an embodiment, described first computing module determines the frame number of the audio coding frame comprised in a transmission frame in the following way:
N=T/t，
Wherein, N represents frame number; T represents the time span of a transmission frame of described digital audio broadcast system; T represents the time span of an audio coding frame of described digital audio broadcast system, is determined by the message sink coding mode of described digital audio broadcast system.
In an embodiment, described second computing module determines the total block data of the channel coding blocks comprised in a transmission frame in the following way:
K=A/b，
Wherein, K represents total block data; A represents the total bytes that a transmission frame of described digital audio broadcast system can transmit, by modulation system and the band bandwidth decision of described digital audio broadcast system; B represents the total bytes of a channel coding blocks, is determined by the channel coding method of described digital audio broadcast system.
In an embodiment, described 4th computing module calculates in a transmission frame total bytes of the N number of audio coding frame comprised in the following way:
B=K*M，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; K represents total block data; M represents the byte number of source coded data in a channel coding blocks.
In an embodiment, described device also comprises:
Hierarchical coding determination module, for determining whether to carry out hierarchical coding according to the modulation system of described digital audio broadcast system or channel coding method; And when hierarchical coding,
Described second computing module further based on the respective channel coding method of Primary layer and enhancement layer and the channel coding blocks block number of modulation system determination Primary layer and the channel coding blocks block number of enhancement layer, and meets:
K=K
b+K
e，
Wherein, K represents total block data; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer;
Described 4th computing module calculates in a transmission frame total bytes of the N number of audio coding frame comprised in the following way:
B=B
b+B
e=K
b*M
b+K
e*M
e，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; B
brepresent the total bytes of the source coded data of Primary layer; B
erepresent the total bytes of the source coded data of enhancement layer; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer; M
brepresent the byte number of source coded data in the channel coding blocks of Primary layer; M
erepresent the byte number of source coded data in the channel coding blocks of enhancement layer.
In an embodiment, when hierarchical coding, described message sink coding module carries out compressed in layers coding to N number of audio coding frame of input continuously in the transmission frame time further, the source coded data total bytes of the Primary layer calculated based on the 4th computing module distributes the Primary layer byte number of each audio coding frame adaptively, and the source coded data total bytes of the enhancement layer calculated based on the 4th computing module distributes the enhancement layer byte number of each audio coding frame adaptively.
The digital audio source coding method of segmental averaging code check of the present invention (with layer and section average bit rate) and device are a kind of channel information source combined coding technology, have following beneficial effect:
(1) for non-layered digital audio message sink coding situation, every frame frame length (or byte number) is distributed adaptively between N number of coded frame of the present invention in each transmission frame, guarantee that audio coder obtains local optimum compression in a transmission frame, thus subjective sound quality in a transmission frame is consistent.
(2) for layering digital audio message sink coding situation, the present invention is difference local optimum compression coding Primary layer and enhancement layer in each transmission frame, obtains better code efficiency.
Therefore, the present invention, can with very little encoder complexity for cost provides higher code efficiency compared with CBR coding method; As compared to ABR with VBR coding method in general sense in digital audio encoding, obviously can reduce the multiplexing complexity of digital audio broadcast system audio coding unit output code flow, also can improve code efficiency simultaneously.
Accompanying drawing explanation
Below in conjunction with drawings and Examples, the invention will be further described, in accompanying drawing:
Fig. 1 is digital audio broadcast system sketch;
Fig. 2 is the flow chart of the digital audio source coding method of the segmental averaging code check of one embodiment of the invention;
Fig. 3 is the flow chart of the digital audio source coding method of the layer and section average bit rate of another embodiment of the present invention;
Fig. 4 is the logic diagram of the digital audio message sink coding device of the segmental averaging code check of one embodiment of the invention;
Fig. 5 is the structural representation of the N frame coded data of non-delamination;
Fig. 6 is the structural representation of the N frame coded data of delamination;
Fig. 7 is the structural representation of the channel coding blocks of the transmission frame of non-delamination;
Fig. 8 is the structural representation of the channel coding blocks of the transmission frame of delamination.
Embodiment
In order to make object of the present invention, technical scheme and advantage clearly understand, below in conjunction with drawings and Examples, the present invention is further elaborated.Should be appreciated that specific embodiment described herein only in order to explain the present invention, be not intended to limit the present invention.
Fig. 1 shows the sketch of digital audio broadcast system 100.As shown in Figure 1, in digital audio broadcast system 100, digital audio is encoded through audio coding unit 101, multiplexing through multiplexer 104 together with data 102, control information 103, then carry out the process such as Channel Coding and Modulation through channel bank 105, then send radio frequency unit 106 to and be transmitted in the air by antenna 107.
As can see from Figure 1, digital audio encoding unit 101 is in whole system foremost, and in general, message sink coding does not need the impact being subject to the parts such as channel, but in order to better coordinate the feature of channel and modulating unit, based on Channel Coding and Modulation parameter, certain constraint is carried out to message sink coding, better code efficiency can be obtained.
The present invention just considers the feature of chnnel coding and modulation technique in digital audio broadcast system and proposes a kind of segmental averaging code check (SegmentedAverageBitRate, and layer and section average bit rate (Layered & SegmentedAverageBitRate S-ABR), LS-ABR) digital audio source coding method, it is a kind of source coding method based on channel, mainly through the code stream of effective control coding end output buffer, ensure that encoder output code flow in Fixed Time Interval (or in a certain constant coding frame number) has a road (corresponding not stratified situation) or multichannel (the corresponding layering number of plies) constant mean bit rate.
Digital audio source coding method of the present invention determines whether to need to carry out hierarchical coding based on the feature of chnnel coding and modulation technique.Specifically, the determination of hierarchical coding of the present invention, in two kinds of situation.
The first situation: determine whether to carry out hierarchical coding according to the modulation system that digital audio broadcast system adopts.If the modulation signal of the modulation system that digital audio broadcast system adopts to input can be divided into multiple different modulation levels; and the data of different brackets can produce the different error rates at receiving terminal; thus there is the effect being similar to and not waiting error protection; now just require that message sink coding provides hierarchical coding ability as far as possible, to be adapted to this modulation technique better.When message sink coding supports layering, then after the process such as chnnel coding, the Primary layer of message sink coding is placed on the high-grade layer modulation of modulation, and enhancement layer is placed on the modulation of inferior grade layer.If the modulation system that digital audio broadcast system adopts is to the as broad as long process of modulation signal of input, then think that modulation does not support hierarchical coding, now message sink coding can not carry out hierarchical coding.
The second situation: whether support that multiple error correction protection determines whether to carry out hierarchical coding according to the channel coding method that digital audio broadcast system adopts.During hierarchical coding, then Primary layer gives high error correction protection class, and enhancement layer gives low protection class.
Fig. 2 shows the flow chart of the digital audio source coding method (S-ABR) 200 of the segmental averaging code check in non-layered situation according to an embodiment of the invention.As shown in Figure 2, this digital audio source coding method 200 comprises the steps:
In step 210, determine the frame number N of the audio coding frame comprised in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system.In digital audio broadcast system, when channel coding method and modulation system are determined, can obtain the time span T of a transmission frame, such as, in CDR system, the time span of a transmission frame is 640mn.Meanwhile, for various different message sink coding pattern, the time span of its each audio coding frame is also determined.Such as adopt DRA coding, namely during high code check, for 48kHz audio signal, time span t=(1024/48) millisecond of each audio coding frame; And for DRA+ coding, namely during low bit-rate, the time span of each audio coding frame is 2 times of DRA frame lengths.Therefore, according to the time span T of a transmission frame of digital audio broadcast system, and the time span t of a digital audio coded frame, the frame number N of the audio coding frame comprised in each transmission frame can be determined, that is:
N=T/t。
Fig. 5 shows the structure chart of N frame coded data in the next transmission frame of non-layered situation, and wherein, parameter B is the total bytes of the N number of audio coding frame in a transmission frame time span, equals the byte number B of each audio coding frame
j(j=1,2 ... N) sum.
In following step 220, determine the total block data K of the channel coding blocks comprised in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system.The total block data K of the channel coding blocks comprised in a transmission frame is that the total amount of data that can be transmitted by a transmission frame and channel coding algorithm are determined.The total bytes of a transmission frame is determined by the modulation system of digital audio broadcast system and band bandwidth.In general, the total bytes of each channel coding blocks (as LDPC coding) is also determined.Pass through the total data byte number of a transmission frame like this divided by the byte number of a channel coding blocks, just can derive the total block data K of the channel coding blocks comprised in a transmission frame, that is:
K=A/b，
Wherein, K represents total block data, and A represents the total bytes that a transmission frame can transmit, and b represents the total bytes of a channel coding blocks.
In following step 230, determine the byte number M of source coded data in a channel coding blocks according to the Channel Coding and Modulation parameter information of digital audio broadcast system.M represents the byte number of protected information in each channel coding blocks, can determine based on channel coding method.Such as, if adopt the LDPC coding of 1/2, then the block length of each LDPC block is 1152 bytes, and in each LDPC block, source coded data part is 1152/2 byte, i.e. M=576 byte.Fig. 7 shows the structure of the channel coding blocks of the next transmission frame of non-layered situation, and the total bytes b due to each channel coding blocks is constant, and source coded data and the channel guard data of each piece should meet:
M+C=b，
Wherein, M represents the byte number of source coded data in a channel coding blocks, and C represents the byte number of channel guard data in a channel coding blocks, and b represents the total bytes of a channel coding blocks.
In following step 240, the total bytes B of the N number of audio coding frame comprised is calculated in a transmission frame, that is: based on the byte number M of source coded data in the channel coding blocks that the total block data K of the channel coding blocks comprised in the transmission frame that step 220 is determined, step 230 are determined
B=K*M。
In following step 250, carry out compressed encoding to the N frame voice data of continuously input in a transmission frame time T, the total bytes B calculated based on step 240 distributes the byte number of each audio coding frame adaptively in N number of audio coding interframe.Namely, according to the difference of every frame audio signal content, distribute the byte number of each frame adaptively, make total coded word joint number be B, guarantee in a transmission frame, obtain local optimum compression, thus subjective sound quality in a transmission frame is consistent.
Fig. 3 shows the flow chart of the digital audio source coding method (LS-ABR) 300 of the layer and section average bit rate under delamination according to an embodiment of the invention.As shown in Figure 3, this digital audio source coding method 300 comprises the steps:
In step 310, determine the frame number N of the audio coding frame comprised in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system.As described in front composition graphs 2, frame number N can determine according to the time span t of the time span T and of of a digital audio broadcast system transmission frame digital audio coded frame, that is:
N=T/t。
Fig. 6 shows the structure chart of N frame coded data in a transmission frame of delamination (Primary layer and enhancement layer), wherein, parameter B is the total bytes of the N number of audio coding frame in a transmission frame time span, equals the Primary layer byte number B of each audio coding frame
jb(j=1,2 ... and enhancement layer byte number B N)
je(j=1,2 ... N) sum.
In following step 320, determine the total block data K of the channel coding blocks comprised in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system.As shown in front composition graphs 2, the total block data K of the channel coding blocks comprised in a transmission frame is that the total amount of data that can be transmitted by a transmission frame and channel coding algorithm are determined, that is:
K=A/b，
Wherein, K represents total block data, and A represents the total bytes that a transmission frame can transmit, and b represents the total bytes of a channel coding blocks.
In following step 330, based on Primary layer and enhancement layer channel coding method separately and the channel coding blocks block number K of modulation system determination Primary layer
bwith the channel coding blocks block number K of enhancement layer
e.For the situation of hierarchical coding, because the total bytes of every layer in a transmission frame may be different, every layer of channel coding method used is different in addition, makes Primary layer and enhancement layer have different channel coding blocks block number K
band K
e, and meet K=K
b+ K
e.
In following step 340, determine the byte number M of source coded data in the channel coding blocks of Primary layer respectively based on the respective channel coding method of Primary layer and enhancement layer and modulation system
bwith the byte number M of source coded data in the channel coding blocks of enhancement layer
e.Such as, if Primary layer adopts 1/4LDPC coding, enhancement layer adopts 1/2LDPC coding, and the block length of each LDPC block is 1152 bytes, then in each LDPC block of Primary layer, source coded data part is 1152/4 byte, i.e. M
b=288 bytes, in each LDPC block of enhancement layer, source coded data part is 1152/2 byte, i.e. M
e=576 bytes.Fig. 8 shows the structure of the channel coding blocks of the next transmission frame of delamination, and the total bytes b due to each channel coding blocks is constant, therefore when layering, and the M of Primary layer channel coding blocks
bbe less than the M of enhancement layer channel coding blocks
e, correspondingly, the trip protection information byte number C of Primary layer channel coding blocks
bbe greater than the trip protection information byte number C of enhancement layer channel coding blocks
e.
In following step 350, calculate the total bytes B of the source coded data of a transmission intra-base-layer in the following way
b, enhancement layer the total bytes B of source coded data
e, and the total bytes B of N number of audio coding frame:
B=B
b+B
e=K
b*M
b+K
e*M
e。
In following step 360, compressed in layers coding is carried out, based on the source coded data total bytes B of the Primary layer that step 350 calculates to the N frame voice data of input continuously in the transmission frame time
bthe Primary layer byte number of each audio coding frame is distributed adaptively, based on the source coded data total bytes B of enhancement layer in N number of audio coding interframe
edistribute the enhancement layer byte number of each audio coding frame adaptively.That is, according to the difference of every frame audio signal content, distribute Primary layer byte number and the enhancement layer byte number of each frame adaptively, make Primary layer total bytes be B
b, enhancement layer total bytes is B
e, thus local optimum compression coding Primary layer and enhancement layer is distinguished in a transmission frame, obtain the code efficiency of equal sign.
In a word, in digital audio broadcast system application, in General Requirements transmission frame, the total bytes of message sink coding is fixed, S-ABR and the LS-ABR coding mode adopting more than the present invention to introduce, can obtain higher code efficiency.
Fig. 4 is the logic diagram of the digital audio message sink coding device 400 of the segmental averaging code check of one embodiment of the invention.As shown in Figure 4, this digital audio message sink coding device 400 comprises the first computing module 410, second computing module 420, the 3rd computing module 430, the 4th computing module 440 and message sink coding module 450, for realizing the digital audio source coding method above described in composition graphs 2 and Fig. 3.Further, this digital audio message sink coding device 400 also includes hierarchical coding determination module, for determining whether to need to carry out hierarchical coding according to the modulation system of digital audio broadcast system or channel coding method.Such as, if the modulation signal of modulation system to input that digital audio broadcast system adopts can be divided into multiple different modulation levels, then layered source can be adopted to encode, to adapt to this modulation technique better.Again such as, if the channel coding method that digital audio broadcast system adopts supports that multiple error correction is protected, then layered source can be adopted to encode, give high error correction protection to Primary layer, give low protection class to enhancement layer.
Further as shown in Figure 4, the frame number N of audio coding frame of the first computing module 410 for determining to comprise in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system.Specifically, the first computing module 410 is by following formula determination frame number N:
N=T/t，
Wherein, N represents frame number; T represents the time span of a transmission frame of described digital audio broadcast system; T represents the time span of an audio coding frame of described digital audio broadcast system, is determined by the message sink coding mode of described digital audio broadcast system.
The total block data K of channel coding blocks of the second computing module 420 for determining to comprise in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system.Specifically, the second computing module 420 determines the total block data K of the channel coding blocks comprised in a transmission frame in the following way:
K=A/b，
Wherein, K represents total block data; A represents the total bytes that a transmission frame of described digital audio broadcast system can transmit, by modulation system and the band bandwidth decision of described digital audio broadcast system; B represents the total bytes of a channel coding blocks, is determined by the channel coding method of described digital audio broadcast system.
When hierarchical coding, the second computing module 420 is further based on Primary layer and enhancement layer channel coding method separately and the channel coding blocks block number K of modulation system determination Primary layer
bwith the channel coding blocks block number K of enhancement layer
e, and meet K=K
b+ K
e.
3rd computing module 430 is for determining the byte number M of source coded data in a channel coding blocks according to the Channel Coding and Modulation parameter information of digital audio broadcast system.M represents the byte number of protected information in each channel coding blocks, can determine based on channel coding method.When hierarchical coding, the 3rd computing module 430 determines the byte number M of source coded data in the channel coding blocks of Primary layer respectively based on the respective channel coding method of Primary layer and enhancement layer and modulation system
bwith the byte number M of source coded data in the channel coding blocks of enhancement layer
e.
4th calculates mould 440 pieces for calculating in a transmission frame total bytes B of the N number of audio coding frame comprised based on the result of the second computing module 420 and the 3rd computing module 430.When non-layered is encoded, the 4th computing module 440 calculates in a transmission frame total bytes B of the N number of audio coding frame comprised in the following way:
B=K*M，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; K represents total block data; M represents the byte number of source coded data in a channel coding blocks.
When hierarchical coding, the 4th computing module 440 calculates in a transmission frame total bytes B of the N number of audio coding frame comprised in the following way:
B=B
b+B
e=K
b*M
b+K
e*M
e，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; B
brepresent the total bytes of the source coded data of Primary layer; B
erepresent the total bytes of the source coded data of enhancement layer; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer; M
brepresent the byte number of source coded data in the channel coding blocks of Primary layer; M
erepresent the byte number of source coded data in the channel coding blocks of enhancement layer.
When non-layered coding, message sink coding module 450 is for carrying out compressed encoding to the N frame voice data of input continuously in the transmission frame time, the total bytes B calculated based on the 4th computing module 440 distributes the byte number of each audio coding frame adaptively in N number of audio coding interframe, namely, according to the difference of every frame audio signal content, distribute the byte number of each frame adaptively, total coded word joint number is made to be B, guarantee in a transmission frame, obtain local optimum compression, thus subjective sound quality in a transmission frame is consistent.
When hierarchical coding, the N number of audio coding frame of message sink coding module 450 to input continuously in the transmission frame time carries out compressed in layers coding, based on the source coded data total bytes B of the Primary layer that the 4th computing module 440 calculates
bdistribute the Primary layer byte number of each audio coding frame adaptively, based on the source coded data total bytes B of the enhancement layer that the 4th computing module 440 calculates
edistribute the enhancement layer byte number of each audio coding frame adaptively, that is, according to the difference of every frame audio signal content, distribute Primary layer byte number and the enhancement layer byte number of each frame adaptively, make Primary layer total bytes be B
b, enhancement layer total bytes is B
e, thus local optimum compression coding Primary layer and enhancement layer is distinguished in a transmission frame, obtain better code efficiency.
Below will with China digital audio broadcasting CDR(ChinaDigitalRadio, Chinese Digital frequency hopping be broadcasted) introduce the embody rule of digital audio source coding method (S-ABR and LS-ABR) of the present invention for example.
In CDR application, digital audio source coding technique adopts China national audio standard DRA and DRA to strengthen coding (DRA+) technology, wherein DRA+ supports low bit rate audio coding and hierarchical coding (comprising stereo and surround sound hierarchical coding), and chnnel coding adopts LDPC, modulation adopts QPSK/QAM and OFDM technology.
Table 1 gives the effective message sink coding code check under different modulating and chnnel coding, if adopt fixed bit rate pattern, then can directly apply, but when adopting variable bit rate pattern, consider the transmittability of a transmission frame, make system and control more complicated.
Effective net load in table 1 different modulating pattern and chnnel coding situation
Major parameter in CDR system is as follows:
In CDR, the time span T=640ms of a transmission frame, if be only modulated to example with transmission mode 1 & 2 and 16QAM, then the gross bit rate of modulation signal is the corresponding total bytes A of 288kbps(is 23040).In CDR, adopt DRA coding, suppose that DRA adopts high code check coding mode, then to time span t=(1024/48) ms of each audio coding frame of 48kHz audio signal.The frame number N=T/t=30 of the audio coding frame therefore comprised in a transmission frame; When using low bit-rate DRA, N=15 frame).
In CDR, chnnel coding adopts LDPC, and the total bytes b of each channel coding blocks is 1152 bytes, therefore comprises 20 encoding blocks, i.e. K=A/b=23040/1152=20 in each transmission frame.
For the situation of non-layered coding, if adopt 1/2LDPC coding, then in each LDPC block, source coded data part is 1152/2 byte, i.e. M=576 byte.Message sink coding packed byte number total in a transmission frame is B=M*K=11520 byte, then the mean bit rate (S-ABR bit rate) of message sink coding is 144kbps.At this moment the meaning that segmental averaging code check S-ABR of the present invention encodes is: in a transmission frame time (640ms), by carrying out DRA compression algorithm to 30 frame pcm audio data of input continuously, total coding byte number is made to be 11520 bytes, wherein, according to the difference of every frame signal content, the byte number of each frame of self-adjusted block, thus obtain local optimum coding.
For the situation of hierarchical coding, if Primary layer adopts 1/4LDPC coding, enhancement layer adopts 1/2LDPC coding, due to the block number K=20 of total channel coding blocks, in order to simplify, assuming that base layer block number K
bwith enhancement layer block number K
eidentical, all equal 10, then the total bytes of a transmission intra-base-layer is B
b=K
b* 1152*(1/4)=2880 bytes, and in a transmission frame, the total bytes of enhancement layer is B
e=K
e* 1152*(1/2)=5760 bytes, message sink coding packed byte number total in a final transmission frame is B=B
b+ B
e=(2880+5760)=8640 byte, the mean bit rate (LS-ABR bit rate) of message sink coding is B*8/0.640=108kbps.At this moment the meaning that layer and section average bit rate LS-ABR of the present invention encodes is: in a transmission frame time (640ms), by carrying out layering DRA compression algorithm to 30 frame pcm audio data of input continuously, Primary layer total bytes is made to be 2880, enhancement layer total bytes is 5760, wherein, according to the Primary layer byte number of each frame of every frame signal content dynamic assignment, according to the enhancement layer byte number of each frame of every frame signal content dynamic assignment, thus obtain local optimum coding.
The foregoing is only preferred embodiment of the present invention, not in order to limit the present invention, all any amendments done within the spirit and principles in the present invention, equivalent replacement and improvement etc., all should be included within protection scope of the present invention.
Claims (8)
1. a digital audio source coding method for segmental averaging code check, is characterized in that, comprise the steps:
S1, determine the frame number N of the audio coding frame comprised in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
S2, determine the total block data of the channel coding blocks comprised in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
S3, determine the byte number of source coded data in a channel coding blocks according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
S4, calculate in a transmission frame total bytes of the N number of audio coding frame comprised based on step S2, S3;
S5, carry out compressed encoding to the N frame voice data of continuously input in the transmission frame time, the total bytes calculated based on step S4 distributes the byte number of each audio coding frame adaptively in N number of audio coding interframe;
Described method also comprises: determine whether to carry out hierarchical coding according to the modulation system of described digital audio broadcast system or channel coding method; And
When hierarchical coding, described step S2 comprises further:
Based on the respective channel coding method of Primary layer and enhancement layer and the channel coding blocks block number of modulation system determination Primary layer and the channel coding blocks block number of enhancement layer, and meet:
K＝K
b+K
e，
Wherein, K represents total block data; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer;
Described step S4 calculates in a transmission frame total bytes of the N number of audio coding frame comprised in the following way:
B＝B
b+B
e＝K
b*M
b+K
e*M
e，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; B
brepresent the total bytes of the source coded data of Primary layer; B
erepresent the total bytes of the source coded data of enhancement layer; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer; M
brepresent the byte number of source coded data in the channel coding blocks of Primary layer; M
erepresent the byte number of source coded data in the channel coding blocks of enhancement layer.
2. method according to claim 1, is characterized in that, described step S1 determines the frame number of the audio coding frame comprised in a transmission frame in the following way:
N＝T/t，
Wherein, N represents frame number; T represents the time span of a transmission frame of described digital audio broadcast system; T represents the time span of an audio coding frame of described digital audio broadcast system, is determined by the message sink coding mode of described digital audio broadcast system.
3. method according to claim 1, is characterized in that, when non-layered is encoded, described step S2 determines the total block data of the channel coding blocks comprised in a transmission frame in the following way:
K＝A/b，
Wherein, K represents total block data; A represents the total bytes that a transmission frame of described digital audio broadcast system can transmit, by modulation system and the band bandwidth decision of described digital audio broadcast system; B represents the total bytes of a channel coding blocks, is determined by the channel coding method of described digital audio broadcast system.
4. method according to claim 1, is characterized in that, when non-layered is encoded, described step S4 calculates in a transmission frame total bytes of the N number of audio coding frame comprised in the following way:
B＝K*M，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; K represents total block data; M represents the byte number of source coded data in a channel coding blocks.
5. method according to claim 1, is characterized in that, when hierarchical coding, described step S5 comprises further:
Compressed in layers coding is carried out to the N frame voice data of input continuously in the transmission frame time, the source coded data total bytes of the Primary layer calculated based on step S4 distributes the Primary layer byte number of each audio coding frame adaptively, and the source coded data total bytes of the enhancement layer calculated based on step S4 distributes the enhancement layer byte number of each audio coding frame adaptively.
6. a digital audio message sink coding device for segmental averaging code check, is characterized in that, comprising:
First computing module, for the frame number N of audio coding frame determining to comprise in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
Second computing module, for the total block data of channel coding blocks determining to comprise in a transmission frame according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
3rd computing module, for determining the byte number of source coded data in a channel coding blocks according to the Channel Coding and Modulation parameter information of digital audio broadcast system;
4th computing module, for calculating in a transmission frame total bytes of the N number of audio coding frame comprised based on the result of the second computing module and the 3rd computing module;
Message sink coding module, for carrying out compressed encoding to the N frame voice data of continuously input in the transmission frame time, the total bytes calculated based on the 4th computing module distributes the byte number of each audio coding frame adaptively in N number of audio coding interframe;
Described device also comprises:
Hierarchical coding determination module, for determining whether to carry out hierarchical coding according to the modulation system of described digital audio broadcast system or channel coding method; And when hierarchical coding,
Described second computing module further based on the respective channel coding method of Primary layer and enhancement layer and the channel coding blocks block number of modulation system determination Primary layer and the channel coding blocks block number of enhancement layer, and meets:
K＝K
b+K
e，
Wherein, K represents total block data; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer;
Described 4th computing module calculates in a transmission frame total bytes of the N number of audio coding frame comprised in the following way:
B＝B
b+B
e＝K
b*M
b+K
e*M
e，
Wherein, B represents in a transmission frame total bytes of the N number of audio coding frame comprised; B
brepresent the total bytes of the source coded data of Primary layer; B
erepresent the total bytes of the source coded data of enhancement layer; K
brepresent the block number of Primary layer; K
erepresent the block number of enhancement layer; M
brepresent the byte number of source coded data in the channel coding blocks of Primary layer; M
erepresent the byte number of source coded data in the channel coding blocks of enhancement layer.
7. device according to claim 6, is characterized in that, described first computing module determines the frame number of the audio coding frame comprised in a transmission frame in the following way:
N＝T/t，
Wherein, N represents frame number; T represents the time span of a transmission frame of described digital audio broadcast system; T represents the time span of an audio coding frame of described digital audio broadcast system, is determined by the message sink coding mode of described digital audio broadcast system.
8. device according to claim 6, it is characterized in that, when hierarchical coding, described message sink coding module carries out compressed in layers coding to the N frame voice data of input continuously in the transmission frame time further, the source coded data total bytes of the Primary layer calculated based on the 4th computing module distributes the Primary layer byte number of each audio coding frame adaptively, and the source coded data total bytes of the enhancement layer calculated based on the 4th computing module distributes the enhancement layer byte number of each audio coding frame adaptively.
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| KR102553316B1 (en) * | 2015-03-06 | 2023-07-10 | 한국전자통신연구원 | Apparatus for generating broadcasting signal frame using layered division multiplexing and method using the same |
| CN107547156A (en) * | 2017-09-04 | 2018-01-05 | 成都德芯数字科技股份有限公司 | Information transfer adaptation method, device and CDR broadcast transmission systems |
| US20190191191A1 (en) * | 2017-12-19 | 2019-06-20 | Western Digital Technologies, Inc. | Hybrid techniques for content distribution with edge devices |
| CN110336644B (en) * | 2019-07-15 | 2020-12-15 | 杭州泽铭睿股权投资有限公司 | Layered coding method under high-dimensional modulation |
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