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CN101107781B - Enhanced performance coding and error correction system and method for legacy communication networks - Google Patents

Enhanced performance coding and error correction system and method for legacy communication networks Download PDF

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CN101107781B
CN101107781B CN2005800443386A CN200580044338A CN101107781B CN 101107781 B CN101107781 B CN 101107781B CN 2005800443386 A CN2005800443386 A CN 2005800443386A CN 200580044338 A CN200580044338 A CN 200580044338A CN 101107781 B CN101107781 B CN 101107781B
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CN101107781A (en
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于燕斌
郑清诒
王纯
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NXP USA Inc
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Mawier International Trade Co Ltd
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L1/00Arrangements for detecting or preventing errors in the information received
    • H04L1/0078Avoidance of errors by organising the transmitted data in a format specifically designed to deal with errors, e.g. location
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Abstract

An encoding and error correction system and method employs an AMR codec (18) by stripping header data from a plurality of legacy system frames (10) having header and traffic channel (TCH) data blocks.Speech data is then encoded using the AMR to create bits for a data block substantially the same as contained in the plurality of frames. The stripped header data is encoded as a long frame header using a fixed convolution coder(24). The speech data is then convolutionally encoded and the long frame header and encoded speech data are combined as a long frame (32). The long frame is then deconstructed into a plurality of equal segments (106, 110) and the segments are transmitted as TCH data in the legacy system frame format.

Description

用于遗留通信网络的增强性能的编码和纠错系统及方法 Enhanced performance coding and error correction system and method for legacy communication networks

技术领域technical field

本发明一般地涉及用于发送系统的编码和纠错领域,更具体而言,涉及带有比特映射和编码的高级声码器(vocoder)的使用,从而改进遗留(legacy)通信系统以增强纠错性能。The present invention relates generally to the field of coding and error correction for transmission systems, and more particularly to the use of advanced vocoders (vocoders) with bit mapping and coding to improve legacy communication systems for enhanced correction wrong performance.

背景技术Background technique

波形源编码和解码(codec)被广泛用于早期的数字移动通信系统中,例如个人手持电话系统(PHS)。由于实现时的技术限制,某些系统设计没有提供适当的信道编码/解码以保护被传输的数据。对于这些系统,当信道质量条件恶化时,为了确保高误码率而使得语音性能无法接受并且某些重要的控制比特容易被破坏。结果,系统中的上层协议和控制机制可能被激活以关断信道。这是在正在进行的通信会话期间丢失连接的最普遍原因之一。Waveform source coding and decoding (codec) is widely used in early digital mobile communication systems, such as Personal Handyphone System (PHS). Due to technical limitations at the time of implementation, some system designs do not provide proper channel coding/decoding to protect the transmitted data. For these systems, speech performance is unacceptable and some important control bits are easily corrupted in order to ensure a high bit error rate when channel quality conditions deteriorate. As a result, upper layer protocols and control mechanisms in the system may be activated to shut down the channel. This is one of the most common reasons for loss of connection during an ongoing communication session.

在某些高级的2G移动系统和所有3G系统中,存在自适应的多速率(AMR)声码器和相应的信道编码能力。在AMR标准下,对于码激励线性预测(CELP)语音编解码器,存在8个不同的数据速率。这些数据速率的范围从12.2kbps到4.75kbps。所传输的语音信息越多,所实现的声音性能就越好。在AMR标准中采用的基本方法是这样的:当信道条件变差时,系统使用具有较低数据传输速率的模式(当然,声音性能较差)。这节省了更多的信道带宽和系统的其他资源来提高比特纠错能力。遗留数字移动系统(例如PHS)中缺乏可比技术是来源于正在研发的算法和与功率和指令速度要求相关的集成电路资源的花费。随着硅技术的出现,数字信号处理器(DSP)在PHS手机中的使用不再是奢侈的元素。In some advanced 2G mobile systems and all 3G systems there is an Adaptive Multi-Rate (AMR) vocoder and corresponding channel coding capability. Under the AMR standard, there are 8 different data rates for the Code Excited Linear Prediction (CELP) speech codec. These data rates range from 12.2kbps to 4.75kbps. The more voice information that is transmitted, the better the sound performance achieved. The basic approach taken in the AMR standard is this: when channel conditions get worse, the system uses a mode with a lower data rate (and, of course, poorer sound performance). This saves more channel bandwidth and other resources of the system to improve bit error correction capability. The lack of comparable technology in legacy digital mobile systems (eg PHS) stems from the expense of algorithms being developed and integrated circuit resources related to power and instruction speed requirements. With the advent of silicon technology, the use of digital signal processors (DSP) in PHS mobile phones is no longer a luxury element.

因此,希望能够在遗留系统中使用AMR声码器能力。Therefore, it is desirable to be able to use AMR vocoder capabilities in legacy systems.

还希望能够通过重新布置比特映射以对某些AMR模式的误比特降低能力提供高达6到7dB增益,来以可以改进早期2G系统的方式应用AMR和纠错。It is also desirable to be able to apply AMR and error correction in a way that could improve early 2G systems by rearranging the bitmap to provide up to a 6 to 7dB gain to the bit error reduction capability of certain AMR modes.

发明内容Contents of the invention

根据本发明的编码和纠错系统与方法通过从多个具有头部和流量信道(TCH)数据块的遗留系统帧中剥离头部数据来利用AMR编解码器的现代益处。语音数据随后利用AMR被编码,以为多个帧中包含的基本相同的数据块创建比特。剥离的头部数据利用固定卷积编码器被编码为长帧头部。语音数据随后被卷积编码,并且长帧头部和经编码的语音数据被组合成长帧。长帧随后被解构成多个相等的分段,并且这些分段作为TCH数据以遗留系统帧格式被发送。The encoding and error correction system and method according to the present invention take advantage of the modern benefits of the AMR codec by stripping header data from multiple legacy system frames with header and traffic channel (TCH) data blocks. The speech data is then encoded using AMR to create bits for substantially identical blocks of data contained in multiple frames. The stripped header data is encoded as a long frame header using a fixed convolutional encoder. The speech data is then convolutionally encoded, and the longframe header and encoded speech data are combined into a longframe. The long frame is then deconstructed into multiple equal segments, and these segments are sent as TCH data in the legacy system frame format.

根据本发明实施例的编码和纠错方法,包括以下步骤:从具有头部和流量信道(TCH)数据块的多个遗留系统帧中剥离头部数据;收集与多个时隙的时段基本相同的时段内的语音数据;将剥离出的头部数据编码成长帧头部;对所述语音数据编码;将所述长帧头部和编码后的语音数据组合成长帧;将所述长帧解构成多个相等的分段;以及将所述分段作为TCH数据以所述遗留系统帧的格式发送。The coding and error correction method according to the embodiment of the present invention comprises the following steps: stripping header data from a plurality of legacy system frames having headers and traffic channel (TCH) data blocks; collecting substantially the same period as a plurality of time slots Speech data in the period of time; Coding the stripped header data into a long frame header; Encoding the speech data; Combining the long frame header and encoded speech data into a long frame; Decoding the long frame forming a plurality of equal segments; and transmitting the segments as TCH data in the legacy system frame format.

其中,所述语音数据的编码包括采用AMR编解码器.所述剥离出的头部数据包括信道标识符和缓慢相关控制信道数据.所述剥离出的头部数据的编码包括使用固定卷积编码器.所述AMR编解码器具有多种模式并且所述方法还包括基于信道质量来选择编解码器模式的步骤.所述卷积编码器是1/2卷积编码器.所述选择模式的步骤包括在BER恶化时在一个类别内向下移动模式,在最低类别极限时跨类别移动,在BER改善时在一个类别内向上移动,以及在最高类别极限时跨类别移动.Wherein, the encoding of the voice data includes using an AMR codec. The stripped header data includes channel identifiers and slow-related control channel data. The encoding of the stripped header data includes using a fixed convolutional code The AMR codec has multiple modes and the method also includes the step of selecting a codec mode based on channel quality. The convolutional encoder is a 1/2 convolutional encoder. The selected mode Steps include moving patterns down within a class when BER deteriorates, across classes at the lowest class limit, up within a class when BER improves, and across classes at the highest class limit.

根据本发明实施例的编码和纠错系统,包括:用于从具有头部和流量信道(TCH)数据块(12a、12b、12c、12d)的多个遗留系统帧中剥离头部数据的装置(20);用于收集与多个时隙的时段基本相同的时段内的语音数据的装置(18);用于将剥离出的头部数据编码成长帧头部的装置(20);用于对所述语音数据编码的装置(24);用于将所述长帧头部和编码后的语音数据组合成长帧(32)的装置;用于将所述长帧解构成多个相等的分段(34a、34b、34c、34d)的装置;以及用于将所述分段作为TCH数据以所述遗留系统帧的格式发送的装置。An encoding and error correction system according to an embodiment of the invention, comprising: means for stripping header data from a plurality of legacy system frames having header and traffic channel (TCH) data blocks (12a, 12b, 12c, 12d) (20); The device (18) that is used to collect the voice data in the time period that is substantially the same as the period of a plurality of time slots; The device (20) that is used to encode the header data that strips out is a long frame head; means (24) for encoding said speech data; means for combining said long frame header and encoded speech data into a long frame (32); for decomposing said long frame into a plurality of equal subdivisions means for segmenting (34a, 34b, 34c, 34d); and means for transmitting said segment as TCH data in said legacy system frame format.

其中,所述语音数据的编码包括采用AMR编解码器(18),其具有多种模式,并且所述系统还包括用于基于信道质量选择编解码器模式的装置。在所述用于选择编解码器模式的装置中,还包括用于在BER恶化时在一个类别内向下移动模式(72、74、76),并且在最低类别极限时跨类别移动(78)的装置,和用于在BER改善时在一个类别内向上移动(80、82、84),并且在最高类别极限时跨类别移动(86)的装置。Wherein, the encoding of the voice data comprises using an AMR codec (18), which has a plurality of modes, and the system further comprises means for selecting a codec mode based on channel quality. In said means for selecting a codec mode, means for moving modes down (72, 74, 76) within a class when BER deteriorates, and across classes (78) at the lowest class limit means, and means for moving up (80, 82, 84) within a class when BER improves, and across classes (86) at the highest class limit.

附图说明Description of drawings

通过结合附图参考以下详细描述,将更好理解本发明的这些和其他特征和优点,在附图中:These and other features and advantages of the present invention will be better understood by referring to the following detailed description when taken in conjunction with the accompanying drawings in which:

图1是现有技术PHS系统流量信道帧和时隙结构的框图;Fig. 1 is the block diagram of prior art PHS system traffic channel frame and time slot structure;

图2是用于构建长帧以解构成用于发送的标准PHS时隙的比特映射和编码的框图;Figure 2 is a block diagram of the bit mapping and encoding used to construct long frames to deconstruct standard PHS slots for transmission;

图3是比特映射后的长帧的框图;Fig. 3 is a block diagram of a long frame after bit mapping;

图4是用于长帧的鲁棒(Robust)AMR流量同步控制信道(RATSCCH)格式的框图;Fig. 4 is the block diagram that is used for the robust (Robust) AMR traffic synchronization control channel (RATSCCH) format of long frame;

图5是用于编码后的数据的交织方案的框图;Figure 5 is a block diagram of an interleaving scheme for coded data;

图6是采用本发明的手机和基站系统的元件的示意图;Fig. 6 is a schematic diagram of elements of a mobile phone and a base station system using the present invention;

图7a是用于本发明的AMR模式以及对模式(Mode)和类别(class)的划分的表;以及Fig. 7 a is the table that is used for the AMR mode of the present invention and the division of mode (Mode) and category (class); And

图7b是本发明采用的使用类别的模式切换算法的流程图。Fig. 7b is a flow chart of the mode switching algorithm using categories adopted by the present invention.

具体实施方式Detailed ways

本发明被限定用于采用PHS通信系统和标准(2G遗留移动系统)的示例性实施例。并入了本发明的技术的PHS系统这里将被称为高级PHS(APHS)。The present invention is defined for an exemplary embodiment employing the PHS communication system and standard (2G legacy mobile system). A PHS system incorporating the technology of the present invention will be referred to herein as an advanced PHS (APHS).

示例性PHS系统的流量信道(TCH)的数据映射如图1所示.PHS是一种时分复用(TDMA)系统.一个帧10长度为5ms并且被划分成8个时隙,其中4个时隙用于上行链路,4个时隙用于下行链路.在每个方向上,三个时隙T1、T2和T3可被用于三个不同的用户,而最后一个时隙是用于所有三个用户的公共控制信道,其在上行链路模式命令和下行链路模式命令C_up和C-down之间交替.The data mapping of the traffic channel (TCH) of an exemplary PHS system is shown in Figure 1. PHS is a time division multiplexing (TDMA) system. A frame 10 is 5 ms in length and is divided into 8 time slots, of which 4 time slots 4 slots are used for uplink and 4 slots are used for downlink. In each direction, three slots T1, T2 and T3 can be used for three different users, while the last slot is used for Common control channel for all three users, which alternates between uplink mode commands and downlink mode commands C_up and C-down.

在图1中,时隙T2在这里被扩展为示例时隙12并被详细论述以说明本发明。区域PR和UW用于物理层的同步。块UW并入了16比特。如随后将更详细论述的,PR和UW在解码器之前被插入发送的帧中以使得它们无法被编码。CI和SA是用于时隙格式信息和连接状态的协议,它们对于连接可靠性而言非常重要。块CI包含4比特,而块SA包含16比特。TCH包含语音数据并由160比特构成。16比特的CRC块提供了检错比特。可以看出,在一帧内,存在160比特的语音数据,因此总声码器速率为160/0.05=32Kbps。这是PHS采用的ADPCM(ITU G.726)的数据速率。In FIG. 1 , time slot T2 is expanded here as an example time slot 12 and discussed in detail to illustrate the invention. Areas PR and UW are used for synchronization of the physical layer. Block UW incorporates 16 bits. As will be discussed in more detail later, PR and UW are inserted into the transmitted frame prior to the decoder so that they cannot be encoded. CI and SA are protocols for slot format information and connection status, which are very important for connection reliability. Block CI contains 4 bits and block SA contains 16 bits. TCH contains voice data and consists of 160 bits. A 16-bit CRC block provides error detection bits. It can be seen that within one frame, there are 160 bits of speech data, so the total vocoder rate is 160/0.05=32Kbps. This is the data rate of ADPCM (ITU G.726) adopted by PHS.

本发明采用AMR编解码器结合比特映射来构建与PHS发送标准兼容的帧,同时添加编码来进行性能增强。在AMR系统中存在8种声码器模式。这8种模式由GSM和3GPP国际标准定义,并且所采用的数据速率为12.2、10.2、7.95、7.4、6.7、5.9、5.15和4.75(Kbps)。这些速率中的每一种都低于PHS系统的32Kbps基本能力,并且允许灵活地格式化将被编码的数据。The present invention adopts AMR codec combined with bit mapping to construct frames compatible with PHS sending standards, and adds coding to enhance performance. There are 8 vocoder modes in the AMR system. These 8 modes are defined by GSM and 3GPP international standards, and the adopted data rates are 12.2, 10.2, 7.95, 7.4, 6.7, 5.9, 5.15 and 4.75 (Kbps). Each of these rates is below the 32Kbps basic capability of the PHS system and allows for flexible formatting of the data to be encoded.

如随后将更详细描述的,本发明提供的源语音和信道编码是在20ms(或4个PHS时隙时间)内完成的。20ms块的交织在人耳敏感度阈值以下。新编码的“帧”这里被称为长帧。为了与PHS标准帧的比特映射兼容,本发明没有利用CI、SA和CRC区域。就是说,编码后的CI和SA数据被放到TCH块中。声码器模式相关消息被与特定信道代码模式编码在一起,并且采用特殊的区域和编码器。信道编码数据块的尺寸越大,结果就越好(误比特能力越高)。最终,在手机和基站之间的连接的最开始,采用用于长帧同步的新控制信道,鲁棒AMR流量同步控制信道(RATSCCH)。在某些特殊情况下,RATSCCH还被用于模式消息。As will be described in more detail later, the source speech and channel coding provided by the present invention is completed within 20 ms (or 4 PHS slot times). The interleaving of 20ms blocks is below the human ear sensitivity threshold. The newly encoded "frames" are referred to herein as long frames. In order to be compatible with the bitmap of the PHS standard frame, the present invention does not utilize CI, SA and CRC areas. That is, coded CI and SA data are put into TCH blocks. Vocoder pattern related messages are coded with specific channel code patterns and employ specific regions and coders. The larger the size of the channel coded data block, the better the result (higher bit error tolerance). Finally, a new control channel for long frame synchronization, the Robust AMR Traffic Synchronization Control Channel (RATSCCH), is adopted at the very beginning of the connection between the handset and the base station. In some special cases, RATSCCH is also used for mode messages.

TCH块被用于语音数据、SA/CI和带内数据。不同的声码器模式具有不同的语音和信道编码参数并且需要不同的数据传输速率。SA/CI和带内信息被编码并且需要恒定的数据速率。APHS中的编码和比特映射如图2所示。TCH blocks are used for voice data, SA/CI and in-band data. Different vocoder modes have different speech and channel coding parameters and require different data transmission rates. SA/CI and in-band information are coded and require a constant data rate. Coding and bit mapping in APHS are shown in Figure 2.

四个PHS时隙12a、12b、12c和12d名义上提供长帧。这四个时隙的CI和SA数据被剥离并组合到APHS长帧头部块14中。名义上,20ms的语音样本16通过AMR声码器18被处理。长帧头部数据通过第一编码器20被处理,并且带内数据22被插入,接着是经AMR处理的语音样本,该语音样本通过卷积的第二编码器24被路由。所产生的长帧如图3所示,其中头部26包含156比特的CI和SA数据,带内块28运载8比特,接着是476比特的编码语音数据块30,从而得到640比特的长帧32。如上所述,该长帧随后被分割成四个160比特长度以插入到4个APHS帧34a、34b、34c和34d的标准TCH块中用于发送。The four PHS slots 12a, 12b, 12c and 12d nominally provide long frames. The CI and SA data of these four slots are stripped and combined into the APHS long frame header block 14 . Nominally, 20 ms speech samples 16 are processed through an AMR vocoder 18 . Long frame header data is processed through a first encoder 20 and in-band data 22 is inserted, followed by AMR processed speech samples which are routed through a second encoder 24 which is convolutional. The resulting long frame is shown in Figure 3, where the header 26 contains 156 bits of CI and SA data, an in-band block 28 carries 8 bits, followed by a 476-bit coded speech data block 30, resulting in a 640-bit long frame 32. As mentioned above, this long frame is then segmented into four 160-bit lengths for insertion into the standard TCH blocks of the four APHS frames 34a, 34b, 34c and 34d for transmission.

在PHS系统中,用于长帧的区域TCH中的数据速率资源为TCH*4=160*4=640比特。为了兼容,每个时隙中包含SA/CI数据的原始区域在APHS中被预留,但是对于APHS中的处理,这些数据被忽略。消息SA/CI与带内消息数据一起被编码并放在原始TCH区域中。In the PHS system, the data rate resource in the regional TCH for the long frame is TCH*4=160*4=640 bits. For compatibility, the raw area containing SA/CI data in each slot is reserved in APHS, but for processing in APHS, these data are ignored. The message SA/CI is encoded together with the in-band message data and placed in the original TCH area.

在所述APHS实施例中,长帧中的最大编码语音数据是476比特。具有不同参数的不同声码器模式和信道编码模式被组合,以生成具有不同尺寸的不同编码语音数据块。如果生成的编码语音数据块大于476比特,则不得不截去某些比特。在这里公开的APHS实施例中,信道编码是利用卷积编码器来实现的。在替换实施例中,采用其他信道编码方法。In the described APHS embodiment, the maximum encoded speech data in a long frame is 476 bits. Different vocoder modes and channel coding modes with different parameters are combined to generate different encoded speech data blocks with different sizes. If the generated coded speech data block is larger than 476 bits, some bits have to be truncated. In the APHS embodiments disclosed herein, channel coding is implemented using a convolutional coder. In alternative embodiments, other channel coding methods are employed.

随后将论述,基于数据的交织,在每个长帧中CI数据只需要被发送一次(等同于每4个PHS帧发送一个).SA数据需要每个时隙被发送.对于SA/CI,所产生的长帧数据如表1所示.It will be discussed later that based on data interleaving, CI data only needs to be sent once in each long frame (equivalent to sending one every 4 PHS frames). SA data needs to be sent every time slot. For SA/CI, the The generated long frame data is shown in Table 1.

表1Table 1

  名称name  SA0SA0   SA1SA1   SA2SA2   SA3SA3   比特数目number of bits  1616   1616   1616   1616

  名称name   CICI   比特数目number of bits   44   CRC比特数目Number of CRC bits   6多项式D<sup>6</sup>+D<sup>5</sup>+D<sup>3</sup>+D<sup>2</sup>+D<sup>1</sup>+16 polynomial D<sup>6</sup>+D<sup>5</sup>+D<sup>3</sup>+D<sup>2</sup>+D<sup>1</sup >+1   用于卷积编码器的输入比特数目The number of input bits for the convolutional encoder   7474   用于1/2卷积编码器的多项式Polynomials for 1/2 convolutional encoders   多项式G0/G0=1G1/G0=1+D+D<sup>3</sup>+D<sup>4</sup>/1+D<sup>3</sup>+D<sup>4</sup>带有拖尾8比特Polynomial G0/G0=1G1/G0=1+D+D<sup>3</sup>+D<sup>4</sup>/1+D<sup>3</sup>+D<sup>4 </sup> with trailing 8 bits   输出比特的数目Number of output bits   148+8=156148+8=156

8比特被用于模式信息,模式信息被映射到长帧的最后8比特,这将在下面示出。8 bits are used for mode information, which is mapped to the last 8 bits of the long frame, which will be shown below.

表2示出编解码器模式和相关的卷积率、输入到卷积编码器的比特数目、从编码器所产生的输出比特数目、CRC和1/2卷积编码器之后的SA比特数目、总比特数目和优选类别。Table 2 shows the codec modes and associated convolution rates, the number of bits input to the convolutional encoder, the number of output bits produced from the encoder, the number of CRC and SA bits after the 1/2 convolutional encoder, Total number of bits and preferred class.

表2Table 2

  编解码器模式codec mode   比率Ratio   输入到卷积编码器的比特数目The number of bits input to the convolutional encoder   从卷积编码器输出的比特数目(应该为476)The number of bits output from the convolutional encoder (should be 476)   CRC和1/2卷积编码器之后的SA比特数目Number of SA bits after CRC and 1/2 convolutional encoder  对于一个块20ms的总比特数目(总共632+8=640)Total number of bits for a block of 20ms (total 632+8=640)   优选类别preferred category   TCH/AFS12.2TCH/AFS12.2   1/21/2   250250   508截去32比特508 truncated 32 bits   156156  632632   1 1   TCH/AFS10.2TCH/AFS10.2   1/31/3   210210   642截去166比特642 truncated to 166 bits   156156  632632   2 2   TCH/AFSTCH/AFS   1/31/3   165165   513513   156156  632632   1 1   7.957.95  截去37比特truncated 37 bits   TCH/AFS7.4TCH/AFS7.4   1/31/3   154154  474474   156156   630630   2 2   TCH/AFS6.7TCH/AFS6.7   1/41/4   140140  576截去100比特576 truncated 100 bits   156156   632632   1 1   TCH/AFS5.9TCH/AFS5.9   1/41/4   124124  520截去44比特520 truncated 44 bits   156156   632632   2 2   TCH/AFS5.15TCH/AFS5.15   1/51/5   109109  565截去89比特565 truncated 89 bits   156156   632632   1 1   TCH/AFS4.75TCH/AFS4.75   1/51/5   101101  535截去59比特535 truncated 59 bits   156156   632632   2 2

如表中所示,SA/CI数据的固定比特数目为156,并且对于四帧,总共有640比特可用于填充在160比特的PHS TCH内,来自语音数据卷积编码器的比特应该总共476比特,并且比特必须被截短来适应长帧。As shown in the table, the fixed number of bits for SA/CI data is 156, and for four frames, a total of 640 bits are available to fill in the 160-bit PHS TCH, and the bits from the speech data convolutional encoder should total 476 bits , and the bits must be truncated to fit the long frame.

对于每种编解码器模式的示例性卷积编码在表3-10中示出,其中定义截去的比特来保持长帧尺寸为640比特。Exemplary convolutional encodings for each codec mode are shown in Table 3-10, where truncated bits are defined to keep the long frame size at 640 bits.

表3table 3

TCH/AFS 12.2编解码器:TCH/AFS 12.2 codec:

250比特{u(0)...u(249)}的块是利用由以下多项式定义的1/2比率卷积编码来编码的:A block of 250 bits {u(0)...u(249)} is encoded using a 1/2 rate convolutional coding defined by the following polynomial:

G0/G0=1G0/G0=1

G1/G0=1+D+D3+D4/1+D3+D4 G1/G0=1+D+D 3 +D 4 /1+D 3 +D 4

产生508个编码的比特{C(0)...C(507)},由以下各项定义:yields 508 coded bits {C(0)...C(507)}, defined by:

r(k)=u(k)+r(k-3)+r(k-4)r(k)=u(k)+r(k-3)+r(k-4)

C(2k)=u(k)C(2k)=u(k)

C(2k+1)=r(k)+r(k-1)+r(k-3)+r(k-4)对于k=0,1,...,249;C(2k+1)=r(k)+r(k-1)+r(k-3)+r(k-4) for k=0,1,...,249;

r(k)=0对于k<0r(k)=0 for k<0

并且(对于编码器的终止):and (for the encoder's termination):

(k)=0(k)=0

C(2k)=r(k-3)+r(k-4)C(2k)=r(k-3)+r(k-4)

C(2k+1)=r(k)+r(k-1)+r(k-3)+r(k-4)对于k=250,251,...,253C(2k+1)=r(k)+r(k-1)+r(k-3)+r(k-4) for k=250,251,...,253

代码是通过以下方式被截短的,即以下32比特不被发送:C(417),C(421),C(425),C(427),C(429),C(433),C(437),C(441),C(443),C(445),C(449),C(453),C(457),C(459),C(461),C(465),C(469),C(473),C(475),C(477),C(481),C(485),C(489),C(491),C(493),C(495),C(497),C(499),C(501),C(503),C(505)和C(507)。The code is truncated in such a way that the following 32 bits are not sent: C(417), C(421), C(425), C(427), C(429), C(433), C( 437), C(441), C(443), C(445), C(449), C(453), C(457), C(459), C(461), C(465), C( 469), C(473), C(475), C(477), C(481), C(485), C(489), C(491), C(493), C(495), C( 497), C(499), C(501), C(503), C(505) and C(507).

表4Table 4

TCH/AFS 10.2编解码器:TCH/AFS 10.2 codec:

210比特{u(0)...u(209)}的块是利用由以下多项式定义的1/3比率卷积编码来编码的:A block of 210 bits {u(0)...u(209)} is encoded using a 1/3 rate convolutional coding defined by the following polynomial:

G1/G3=1+D+D3+D4/1+D+D2+D3+D4 G1/G3=1+D+D 3 +D 4 /1+D+D 2 +D 3 +D 4

G2/G3=1+D2+D4/1+D+D2+D3+D4 G2/G3=1+D 2 +D 4 /1+D+D 2 +D 3 +D 4

G3/G3=1G3/G3=1

从而产生642个编码的比特{C(0)...C(641)},由以下各项定义:This results in 642 coded bits {C(0)...C(641)}, defined by:

r(k)=0r(k)=0

C(3k)=r(k)+r(k-1)+r(k-3)+r(k-4)C(3k)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(3k+1)=r(k)+r(k-2)+r(k-4)C(3k+1)=r(k)+r(k-2)+r(k-4)

C(3k+2)=r(k-1)+r(k-2)+r(k-3)+r(k-4)对于k=210,211,...,213C(3k+2)=r(k-1)+r(k-2)+r(k-3)+r(k-4) for k=210,211,...,213

代码是通过以下方式被截短的,即以下22比特不被发送:C(1),C(4),C(7),C(10),C(16),C(19),C(22),C(28),C(31),C(34),C(40),C(43),C(46),C(52),C(55),C(58),C(64),C(67),C(70),C(76),C(79)和C(82)。所有这些操作将产生620比特。The codes are truncated in such a way that the following 22 bits are not sent: C(1), C(4), C(7), C(10), C(16), C(19), C( 22), C(28), C(31), C(34), C(40), C(43), C(46), C(52), C(55), C(58), C( 64), C(67), C(70), C(76), C(79) and C(82). All these operations will result in 620 bits.

代码是通过以下方式被截短的,即以下166比特不被发送:C(1),C(4),C(7),C(10),C(16),C(19),C(22),C(28),C(31),C(34),C(40),C(43),C(46),C(52),C(55),C(58),C(64),C(67),C(70),C(76),C(79),C(82),C(88),C(91),C(94),C(100),C(103),C(106),C(112),C(115),C(118),C(124),C(127),C(130),C(136),C(139),C(142),C(148),C(208),C(211),C(214),C(220),C(223),C(226),C(232),C(235),C(238),C(244),C(247),C(250),C(256),C(259),C(151),C(154),C(160),C(163),C(166),C(172),C(175),C(178),C(184),C(187),C(190),C(196),C(199),C(202),C(262),C(268),C(271),C(274),C(280),C(283),C(286),C(292),C(295),C(298),C(304),C(307),C(310),C(316),C(319),C(322),C(325),C(328),C(331),C(334),C(337),C(340),C(343),C(346),C(349),C(352),C(355),C(358),C(361),C(364),C(367),C(370),C(373),C(376),C(379),C(382),C(385),C(388),C(391),C(394),C(397),C(400),C(403),C(406),C(409),C(412),C(415),C(418),C(421),C(424),C(427),C(430),C(433),C(436),C(439),C(442),C(445),C(448),C(451),C(454),C(457),C(460),C(463),C(466),C(469),C(472),C(475),C(478),C(481),C(484),C(487),C(490),C(493),C(496),C(499),C(502),C(505),C(508),C(511),C(514),C(517),C(520),C(523),C(526),C(529),C(532),C(535),C(538),C(541),C(544),C(547),C(550),C(553)和C(556).The codes are truncated in such a way that the following 166 bits are not sent: C(1), C(4), C(7), C(10), C(16), C(19), C( 22), C(28), C(31), C(34), C(40), C(43), C(46), C(52), C(55), C(58), C( 64), C(67), C(70), C(76), C(79), C(82), C(88), C(91), C(94), C(100), C( 103), C(106), C(112), C(115), C(118), C(124), C(127), C(130), C(136), C(139), C( 142), C(148), C(208), C(211), C(214), C(220), C(223), C(226), C(232), C(235), C( 238), C(244), C(247), C(250), C(256), C(259), C(151), C(154), C(160), C(163), C( 166), C(172), C(175), C(178), C(184), C(187), C(190), C(196), C(199), C(202), C( 262), C(268), C(271), C(274), C(280), C(283), C(286), C(292), C(295), C(298), C( 304), C(307), C(310), C(316), C(319), C(322), C(325), C(328), C(331), C(334), C( 337), C(340), C(343), C(346), C(349), C(352), C(355), C(358), C(361), C(364), C( 367), C(370), C(373), C(376), C(379), C(382), C(385), C(388), C(391), C(394), C( 397), C(400), C(403), C(406), C(409), C(412), C(415), C(418), C(421), C(424), C( 427), C(430), C(433), C(436), C(439), C(442), C(445), C(448), C(451), C(454), C( 457), C(460), C(463), C(466), C(469), C(472), C(475), C(478), C(481), C(484), C( 487), C(490), C(493), C(496), C(499), C(502), C(505), C(508), C(511), C(514), C(517), C(520), C(523), C(526), C(529), C(532), C(535), C(538), C(541), C(544), C(547), C(550), C(553) and C(556).

表5table 5

TCH/AFS 7.95编解码器:TCH/AFS 7.95 codec:

165比特{u(0)...u(164)}的块是利用由以下多项式定义的1/3比率卷积编码来编码的:A block of 165 bits {u(0)...u(164)} is encoded using a 1/3 rate convolutional coding defined by the following polynomial:

G4/G4=1G4/G4=1

G5/G4=1+D+D4+D6/1+D2+D3+D5+D6 G5/G4=1+D+D 4 +D 6 /1+D 2 +D 3 +D 5 +D 6

G6/G4=1+D+D2+D3+D4+D6/1+D2+D3+D5+D6 G6/G4=1+D+D 2 +D 3 +D 4 +D 6 /1+D 2 +D 3 +D 5 +D 6

从而产生513个编码的比特{C(0)...C(512)},由以下各项定义:This results in 513 coded bits {C(0)...C(512)}, defined by:

r(k)=u(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)r(k)=u(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)

C(3k)=u(k)C(3k)=u(k)

C(3k+1)=r(k)+r(k-1)+r(k-4)+r(k-6)C(3k+1)=r(k)+r(k-1)+r(k-4)+r(k-6)

C(3k+2)=r(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)C(3k+2)=r(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)

对于k=0,1,...,164For k=0,1,...,164

r(k)=0  对于k<0r(k)=0 for k<0

并且(对于编码器的终止):and (for the encoder's termination):

r(k)=0r(k)=0

C(3k)=r(k-2)+r(k-3)+r(k-5)+r(k-6)C(3k)=r(k-2)+r(k-3)+r(k-5)+r(k-6)

C(3k+1)=r(k)+r(k-1)+r(k-4)+r(k-6)C(3k+1)=r(k)+r(k-1)+r(k-4)+r(k-6)

C(3k+2)=r(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)  对于k=165,166,...,170C(3k+2)=r(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6) for k=165 ,166,...,170

代码是通过以下方式被截短的,即以下37比特不被发送:C(1),C(2),C(4),C(5),C(8),C(22),C(70),C(118),C(166),C(214),C(262),C(310),C(317),C(319),C(325),C(332),C(334),C(341),C(343),C(349),C(356),C(358),C(365),C(367),C(373),C(380),C(382),C(385),C(389),C(391),C(397),C(404),C(406),C(409),C(413),C(415),和C(512)。The codes are truncated in such a way that the following 37 bits are not transmitted: C(1), C(2), C(4), C(5), C(8), C(22), C( 70), C(118), C(166), C(214), C(262), C(310), C(317), C(319), C(325), C(332), C( 334), C(341), C(343), C(349), C(356), C(358), C(365), C(367), C(373), C(380), C( 382), C(385), C(389), C(391), C(397), C(404), C(406), C(409), C(413), C(415), and C (512).

表6Table 6

TCH/AFS 7.4编解码器:TCH/AFS 7.4 codec:

154比特{u(0)...u(153)}的块是利用由以下多项式定义的1/3比率卷积编码来编码的:A block of 154 bits {u(0)...u(153)} is encoded using a 1/3 rate convolutional coding defined by the following polynomial:

G1/G3=1+D+D3+D4/1+D+D2+D3+D4 G1/G3=1+D+D 3 +D 4 /1+D+D 2 +D 3 +D 4

G2/G3=1+D2+D4/1+D+D2+D3+D4 G2/G3=1+D 2 +D 4 /1+D+D 2 +D 3 +D 4

G3/G3=1G3/G3=1

从而产生474个编码的比特{C(0)...C(473)},由以下各项定义:This results in 474 coded bits {C(0)...C(473)}, defined by:

r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)

C(3k)=r(k)+r(k-1)+r(k-3)+r(k-4)C(3k)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(3k+1)=r(k)+r(k-2)+r(k-4)C(3k+1)=r(k)+r(k-2)+r(k-4)

C(3k+2)=u(k)    对于k=0,1,...,153C(3k+2)=u(k) For k=0,1,...,153

并且(对于编码器的终止):and (for the encoder's termination):

r(k)=0r(k)=0

C(3k)=r(k)+r(k-1)+r(k-3)+r(k-4)C(3k)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(3k+1)=r(k)+r(k-2)+r(k-4)C(3k+1)=r(k)+r(k-2)+r(k-4)

C(3k+2)=r(k-1)+r(k-2)+r(k-3)+r(k-4)  对于k=154,155,...,157C(3k+2)=r(k-1)+r(k-2)+r(k-3)+r(k-4) for k=154, 155, ..., 157

表7Table 7

TCH/AFS 6.7编解码器:TCH/AFS 6.7 codec:

140比特{u(0)...u(139)}的块是利用由以下多项式定义的1/4比率卷积编码来编码的:A block of 140 bits {u(0)...u(139)} is encoded using a 1/4 rate convolutional coding defined by the following polynomial:

G1/G3=1+D+D3+D4/1+D+D2+D3+D4 G1/G3=1+D+D 3 +D 4 /1+D+D 2 +D 3 +D 4

G2/G3=1+D2+D4/1+D+D2+D3+D4 G2/G3=1+D 2 +D 4 /1+D+D 2 +D 3 +D 4

G3/G3=1G3/G3=1

G3/G3=1G3/G3=1

从而产生576个编码的比特{C(0)...C(575)},由以下各项定义:This results in 576 coded bits {C(0)...C(575)}, defined by:

r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)

C(4k)=r(k)+r(k-1)+r(k-3)+r(k-4)C(4k)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(4k+1)=r(k)+r(k-2)+r(k-4)C(4k+1)=r(k)+r(k-2)+r(k-4)

C(4k+2)=u(k)C(4k+2)=u(k)

C(4k+3)=u(k)对于k=0,1,....139C(4k+3)=u(k) for k=0,1,....139

r(k)=0  对于k<0r(k)=0 for k<0

而且(对于编码器的终止):And (for the encoder's termination):

r(k)=0r(k)=0

C(4k)=r(k)+r(k-1)+r(k-3)+r(k-4)C(4k)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(4k+1)=r(k)+r(k-2)+r(k-4)C(4k+1)=r(k)+r(k-2)+r(k-4)

C(4k+2)=r(k-1)+r(k-2)+r(k-3)+r(k-4)C(4k+2)=r(k-1)+r(k-2)+r(k-3)+r(k-4)

C(4k+3)=r(k-1)+r(k-2)+r(k-3)+r(k-4)  对于k=140,141,...,143C(4k+3)=r(k-1)+r(k-2)+r(k-3)+r(k-4) for k=140,141,...,143

代码是通过以下方式被截短的,即以下100比特不被发送:C(1),C(3),C(7),C(11),C(15),C(27),C(39),C(55),C(67),C(79),C(95),C(107),C(119),C(135),C(147),C(159),C(175),C(187),C(199),C(215),C(227),C(239),C(255),C(267),C(279),C(287),C(291),C(295),C(299),C(303),C(307),C(311),C(315),C(319),C(323),C(327),C(331),C(335),C(339),C(343),C(347),C(351),C(355),C(359),C(363),C(367),C(369),C(371),C(375),C(377),C(379),C(383),C(385),C(387),C(391),C(393),C(395),C(399),C(401),C(403),C(407),C(409),C(411),C(415),C(417),C(419),C(423),C(425),C(427),C(431),C(433),C(435),C(439),C(441),C(443),C(447),C(449),C(451),C(455),C(457),C(459),C(463),C(465),C(467),C(471),C(473),C(475),C(479),C(481),C(483),C(487),C(489),C(491),C(495),C(497),C(499),C(5O3),C(505),C(507)和C(511).The codes are truncated in such a way that the following 100 bits are not sent: C(1), C(3), C(7), C(11), C(15), C(27), C( 39), C(55), C(67), C(79), C(95), C(107), C(119), C(135), C(147), C(159), C( 175), C(187), C(199), C(215), C(227), C(239), C(255), C(267), C(279), C(287), C( 291), C(295), C(299), C(303), C(307), C(311), C(315), C(319), C(323), C(327), C( 331), C(335), C(339), C(343), C(347), C(351), C(355), C(359), C(363), C(367), C( 369), C(371), C(375), C(377), C(379), C(383), C(385), C(387), C(391), C(393), C( 395), C(399), C(401), C(403), C(407), C(409), C(411), C(415), C(417), C(419), C( 423), C(425), C(427), C(431), C(433), C(435), C(439), C(441), C(443), C(447), C( 449), C(451), C(455), C(457), C(459), C(463), C(465), C(467), C(471), C(473), C( 475), C(479), C(481), C(483), C(487), C(489), C(491), C(495), C(497), C(499), C( 5O3), C(505), C(507) and C(511).

表8Table 8

TCH/AFS 5.9编解码器:TCH/AFS 5.9 codec:

124比特{u(0)...u(123)}的块是利用由以下多项式定义的1/4比率卷积编码来编码的:A block of 124 bits {u(0)...u(123)} is encoded using a 1/4 rate convolutional coding defined by the following polynomial:

G4/G6=1+D2+D3+D5+D6/1+D+D2+D3+D4+D6 G4/G6=1+D 2 +D 3 +D 5 +D 6 /1+D+D 2 +D 3 +D 4 +D 6

G5/G6=1+D+D4+D6/1+D+D2+D3+4+D6 G5/G6=1+D+D 4 +D 6 /1+D+D 2 +D 3 + 4 +D 6

G6/G6=1G6/G6=1

G6/G6=1G6/G6=1

从而产生520个编码的比特{C(0)...C(519)},由以下各项定义:This results in 520 coded bits {C(0)...C(519)}, defined by:

r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)

C(4k)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)C(4k)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)

C(4k+1)=r(k)+r(k-1)+r(k-4)+r(k-6)C(4k+1)=r(k)+r(k-1)+r(k-4)+r(k-6)

C(4k+2)=u(k)C(4k+2)=u(k)

C(4k+3)=u(k)   对于k=0,1,....,123C(4k+3)=u(k) For k=0, 1, ..., 123

r(k)=0   对于k<0r(k)=0 for k<0

而且(对于编码器的终止):And (for the encoder's termination):

r(k)=0r(k)=0

C(4k)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)C(4k)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)

C(4k+1)=r(k)+r(k-1)+r(k-4)+r(k-6)C(4k+1)=r(k)+r(k-1)+r(k-4)+r(k-6)

C(4k+2)=r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)C(4k+2)=r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)

C(4k+3)=r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)  对于k=124,125,...,129C(4k+3)=r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6) For k=124, 125, .. ., 129

代码是通过以下方式被截短的,即以下44比特不被发送:C(0),C(1),C(3),C(5),C(7),C(11),C(15),C(31),C(47),C(63),C(79),C(95),C(111),C(127),C(143),C(159),C(175),C(191),C(207),C(223),C(239),C(255),C(271),C(287),C(303),C(319),C(327),C(331),C(335),C(343),C(347),C(351),C(359),C(363),C(367),C(375),C(379),C(383),C(391),C(395),C(399),C(407),C(411)和C(415)。The codes are truncated in such a way that the following 44 bits are not sent: C(0), C(1), C(3), C(5), C(7), C(11), C( 15), C(31), C(47), C(63), C(79), C(95), C(111), C(127), C(143), C(159), C( 175), C(191), C(207), C(223), C(239), C(255), C(271), C(287), C(303), C(319), C( 327), C(331), C(335), C(343), C(347), C(351), C(359), C(363), C(367), C(375), C( 379), C(383), C(391), C(395), C(399), C(407), C(411) and C(415).

表9Table 9

TCH/AFS 5.15编解码器:TCH/AFS 5.15 codec:

109比特{u(0)...u(108)}的块是利用由以下多项式定义的1/5比率卷积编码来编码的:A block of 109 bits {u(0)...u(108)} is encoded using a 1/5 rate convolutional coding defined by the following polynomial:

G1/G3=1+D+D3+D4/1+D+D2+D3+D4 G1/G3=1+D+D 3 +D 4 /1+D+D 2 +D 3 +D 4

G1/G3=1+D+D3+D4/1+D+D2+D3+D4 G1/G3=1+D+D 3 +D 4 /1+D+D 2 +D 3 +D 4

G2/G3=1+D2+D4/1+D+D2+D3+D4 G2/G3=1+D 2 +D 4 /1+D+D 2 +D 3 +D 4

G3/G3=1G3/G3=1

G3/G3=1G3/G3=1

从而产生565个编码的比特{C(0)...C(564)},由以下各项定义:This results in 565 coded bits {C(0)...C(564)}, defined by:

r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)

C(5k)=r(k)+r(k-1)+r(k-3)+r(k-4)C(5k)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(5k+1)=r(k)+r(k-1)+r(k-3)+r(k-4)C(5k+1)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(5k+2)=r(k)+r(k-2)+r(k-4)C(5k+2)=r(k)+r(k-2)+r(k-4)

C(5k+3)=u(k)C(5k+3)=u(k)

C(5k+4)=u(k)   对于k=0,1,...,108C(5k+4)=u(k) For k=0,1,...,108

r(k)=0     对于k<0r(k)=0 for k<0

而且(对于编码器的终止):And (for the encoder's termination):

r(k)=0r(k)=0

C(5k)=r(k)+r(k-1)+r(k-3)+r(k-4)C(5k)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(5k+1)=r(k)+r(k-1)+r(k-3)+r(k-4)C(5k+1)=r(k)+r(k-1)+r(k-3)+r(k-4)

C(5k+2)=r(k)+r(k-2)+r(k-4)C(5k+2)=r(k)+r(k-2)+r(k-4)

C(5k+3)=r(k-1)+r(k-2)+r(k-3)+r(k-4)C(5k+3)=r(k-1)+r(k-2)+r(k-3)+r(k-4)

C(5k+4)=r(k-1)+r(k-2)+r(k-3)+r(k-4)  对于k=109,110,...,112C(5k+4)=r(k-1)+r(k-2)+r(k-3)+r(k-4) for k=109,110,...,112

代码是通过以下方式被截短的,即以下89比特不被发送:C(0),C(4),C(5),C(9),C(10),C(14),C(15),C(20),C(25),C(30),C(35),C(40),C(50),C(60),C(70),C(80),C(90),C(100),C(110),C(120),C(13O),C(140),C(150),C(160),C(170),C(180),C(190),C(200),C(210),C(220),C(230),C(240),C(250),C(260),C(270),C(280),C(290),C(300),C(310),C(315),C(320),C(325),C(330),C(334),C(335),C(340),C(344),C(345),C(350),C(354),C(355),C(360),C(364),C(365),C(370),C(374),C(375),C(380),C(384),C(385),C(390),C(394),C(395),C(400),C(404),C(405),C(410),C(414),C(415),C(420),C(424),C(425),C(430),C(434),C(435),C(440),C(444),C(445),C(450),C(454),C(455),C(460),C(464),C(465),C(470),C(474),C(475),C(480)和C(484)。The codes are truncated in such a way that the following 89 bits are not transmitted: C(0), C(4), C(5), C(9), C(10), C(14), C( 15), C(20), C(25), C(30), C(35), C(40), C(50), C(60), C(70), C(80), C( 90), C(100), C(110), C(120), C(13O), C(140), C(150), C(160), C(170), C(180), C( 190), C(200), C(210), C(220), C(230), C(240), C(250), C(260), C(270), C(280), C( 290), C(300), C(310), C(315), C(320), C(325), C(330), C(334), C(335), C(340), C( 344), C(345), C(350), C(354), C(355), C(360), C(364), C(365), C(370), C(374), C( 375), C(380), C(384), C(385), C(390), C(394), C(395), C(400), C(404), C(405), C( 410), C(414), C(415), C(420), C(424), C(425), C(430), C(434), C(435), C(440), C( 444), C(445), C(450), C(454), C(455), C(460), C(464), C(465), C(470), C(474), C( 475), C(480) and C(484).

表10Table 10

TCH/AFS 4.75编解码器:TCH/AFS 4.75 codec:

101比特{u(0)...u(100)}的块是利用由以下多项式定义的1/5比率卷积编码来编码的:A block of 101 bits {u(0)...u(100)} is encoded using a rate 1/5 convolutional coding defined by the following polynomial:

G4/G6=1+D2+D3+D5+D6/1+D+D2+D3+D4+D6 G4/G6=1+D 2 +D 3 +D 5 +D 6 /1+D+D 2 +D 3 +D 4 +D 6

G4/G6=1+D2+D3+D5+D6/1+D+D2+D3+D4+D6 G4/G6=1+D 2 +D 3 +D 5 +D 6 /1+D+D 2 +D 3 +D 4 +D 6

G5/G6=1+D+D4+D6/1+D+D2+D3+D4+D6 G5/G6=1+D+D 4 +D 6 /1+D+D 2 +D 3 +D 4 +D 6

G6/G6=1G6/G6=1

G6/G6=1G6/G6=1

从而产生535个编码的比特{C(0)...C(534)},由以下各项定义:This results in 535 coded bits {C(0)...C(534)}, defined by:

r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)r(k)=u(k)+r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)

C(5k)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)C(5k)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)

C(5k+1)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)C(5k+1)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)

C(5k+2)=r(k)+r(k-1)+r(k-4)+r(k-6)C(5k+2)=r(k)+r(k-1)+r(k-4)+r(k-6)

C(5k+3)=u(k)C(5k+3)=u(k)

C(5k+4)=u(k)   对于k=0,1,...,100C(5k+4)=u(k) for k=0,1,...,100

r(k)=0  对于k<0r(k)=0 for k<0

而且(对于编码器的终止):And (for the encoder's termination):

r(k)=0r(k)=0

C(5k)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)C(5k)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)

C(5k+1)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)C(5k+1)=r(k)+r(k-2)+r(k-3)+r(k-5)+r(k-6)

C(5k+2)=r(k)+r(k-1)+r(k-4)+r(k-6)C(5k+2)=r(k)+r(k-1)+r(k-4)+r(k-6)

C(5k+3)=r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)C(5k+3)=r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)

C(5k+4)=r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6)  对于k=101,102,...,106C(5k+4)=r(k-1)+r(k-2)+r(k-3)+r(k-4)+r(k-6) For k=101, 102, .. ., 106

代码是通过以下方式被截短的,即以下59比特不被发送:C(0),C(5),C(15),C(25),C(35),C(45),C(55),C(65),C(75),C(85),C(95),C(105),C(115),C(125),C(135),C(145),C(155),C(165),C(175),C(185),C(195),C(205),C(215),C(225),C(235),C(245),C(255),C(265),C(275),C(285),C(295),C(305),C(315),C(325),C(335),C(345),C(355),C(365),C(375),C(385),C(395),C(400),C(405),C(410),C(415),C(420),C(425),C(430),C(435),C(440),C(445),C(450),C(455),C(459),C(460),C(465),C(470),C(475)和C(479)。The code is truncated in such a way that the following 59 bits are not sent: C(0), C(5), C(15), C(25), C(35), C(45), C( 55), C(65), C(75), C(85), C(95), C(105), C(115), C(125), C(135), C(145), C( 155), C(165), C(175), C(185), C(195), C(205), C(215), C(225), C(235), C(245), C( 255), C(265), C(275), C(285), C(295), C(305), C(315), C(325), C(335), C(345), C( 355), C(365), C(375), C(385), C(395), C(400), C(405), C(410), C(415), C(420), C( 425), C(430), C(435), C(440), C(445), C(450), C(455), C(459), C(460), C(465), C( 470), C(475) and C(479).

RATSCCH长帧在图4所示TCH块中采用不同格式。RATSCCH在两种情况下被使用。在手机和基站之间连接的最开始,其被用于长帧同步。而且,在某些困难情况下,RATSCCH中的带内消息被用于将编码器模式信息与常规帧的带内消息一起提供。在本发明中使用的RATSCCH可与GSM/3G系统中广泛用于告知PS/CS改变AMR模式和类别的格式相比拟。The RATSCCH long frame adopts different formats in the TCH block shown in FIG. 4 . RATSCCH is used in two cases. It is used for long frame synchronization at the very beginning of the connection between the mobile phone and the base station. Also, in some difficult cases, in-band messages in the RATSCCH are used to provide encoder mode information together with the in-band messages of regular frames. The RATSCCH used in the present invention can be compared with the format widely used in the GSM/3G system to notify the PS/CS to change the AMR mode and category.

在一个实施例中,标准PHS 5ms突发中用于发送的长帧数据的交织按图5所示方式实现。长帧的比特按表11所示方式被交织。In one embodiment, the interleaving of the long frame data for sending in the standard PHS 5ms burst is implemented in the manner shown in FIG. 5 . The bits of the long frame are interleaved as shown in Table 11.

表11Table 11

Figure G2005800443386D00161
Figure G2005800443386D00161

为了交织,比特被分割成偶数和奇数比特102、104,并根据该表被交织。在108获得80比特数据的八个分段106,其随后与来自之前20ms分段110的八个分段相交织。在交织之后,在PHS时隙的TCH块中一次发送两个分段。For interleaving, the bits are split into even and odd bits 102, 104 and interleaved according to the table. Eight segments 106 of 80-bit data are obtained at 108 , which are then interleaved with eight segments from the preceding 20 ms segment 110 . After interleaving, two segments are sent at a time in a TCH block of a PHS slot.

在常规数据传输模式中,系统以与GSM和3GPP系统中的AMR类似的方式工作.如图6所示,在手机36中,用于输入语音的语音编码器38对数据采样,并将编码后的数据提供到信道编码器40,该信道编码器40将长帧转换成PHS格式5ms帧以用于发送.上行链路语音数据随后被发送到基站42,在基站42中,信道解码器44将PHS标准帧拆解成长帧格式,该长帧格式随后被传递到语音解码器46.在解码期间,由Viterbi解码器在卷积解码过程中估计误比特和相应的SNR(相关信道条件).当在某一时段中进行估计之后,编解码器适配单元48对是否应该在接收方向上改变编码模式进行判决(随后将对此进行详细描述).请求消息(上行链路节点命令50)被插入到带内区域中并被传输到手机.In the conventional data transmission mode, the system works in a similar manner to AMR in GSM and 3GPP systems. As shown in Figure 6, in the mobile phone 36, the voice encoder 38 for inputting voice samples the data and encodes The data of the channel encoder 40 is provided to the channel encoder 40, which converts the long frame into a PHS format 5ms frame for transmission. The uplink voice data is then sent to the base station 42, and in the base station 42, the channel decoder 44 will The PHS standard frame is disassembled into a long frame format, which is then passed to the speech decoder 46. During decoding, bit errors and corresponding SNRs (relative channel conditions) are estimated by the Viterbi decoder during convolutional decoding. When After making an estimate in a certain period, the codec adaptation unit 48 decides whether the coding mode should be changed in the receive direction (this will be described in detail later). The request message (uplink node command 50) is inserted into the in-band area and transmitted to the mobile phone.

从基站到手机的下行链路数据的操作可与用于输入语音的语音编码器52编码的语音数据相比拟,其采样数据并将编码后的数据提供到信道编码器54,该信道编码器54将长帧转换成PHS格式5ms帧以用于发送。下行链路语音数据随后被发送到手机,在手机中,信道解码器56将PHS标准帧拆解成长帧格式,该长帧格式随后被传递到语音解码器58。在解码期间,估计误比特和相应的SNR(相关信道条件)。当在某一时段中进行估计之后,编解码器适配单元60对是否应该在接收方向上改变编码模式进行判决。请求消息(下行链路节点命令62)被插入到带内区域中并被传输到基站。The operation of the downlink data from the base station to the handset is comparable to speech data encoded by a speech encoder 52 for incoming speech which samples the data and provides the encoded data to a channel encoder 54 which Convert long frames to PHS format 5ms frames for sending. The downlink voice data is then sent to the handset where the channel decoder 56 breaks down the PHS standard frame into a long frame format which is then passed to the speech decoder 58 . During decoding, bit errors and corresponding SNRs (relative channel conditions) are estimated. After making an estimate in a certain period, the codec adaptation unit 60 makes a decision as to whether the coding mode should be changed in the receive direction. The request message (downlink node command 62) is inserted into the in-band area and transmitted to the base station.

对于每个发送的长帧,发送器的编码模式(索引)是基于模式命令来确定的,所述模式命令基于图7a和7b所示类别利用算法来移动模式。PHS标准要求允许简化的算法用于标准GSM/3G系统上的模式调整。相邻模式处于不同的类别中,所述类别有效地分割模式总数,如图7a中的表所示。如图7b所示,如果发送在模式11和类别1(12.2kbps)中进行,并且来自卷积解码器的差错估计指示需要降低比特率,模式命令则使得同一类别中的一种模式向下移动72,即移动到模式10类别1(7.95kbps)。如果BER仍旧太高,则命令第二次下移74到模式01类别1(6.7kbps)。BER仍太高导致再次下移76到模式00类别1(5.15kbps)。如果BER仍然太高,则执行最后的跨类移动78到模式00类别2(4.74kbps)。因此,可用模式的整个范围跨越4次命令移动。类似地,如果BER被减小到预定阈值以下,命令则上移类别以增大发送速度。例如,如果系统已经处于最低模式中,即处于模式00类别2(4.74kbps)中,并且BER现在得到改善,命令则向上移动80模式到模式01类别2(5.9kbps),而不是返回模式00类别1。进一步的改善致使移动82到模式10类别2(7.4kbps),再进一步的改善致使移动84到模式11类别2(10.2kbps)。如果可获得进一步的改善,算法则致使模式跨类移动86,即移动到模式11类别1(12.2kbps)以用于最高发送速度。BER在类别范围中间的改善或恶化致使在类别内向上或向下移动,如移动88和90所示出的。该算法允许基于类别内模式的循环移动方式,对类别的移动仅仅发生在到达类别中的最小或最大发送能力时。For each long frame transmitted, the encoding mode (index) of the transmitter is determined based on the mode command using an algorithm to shift the mode based on the classes shown in Figures 7a and 7b. The PHS standard requirements allow a simplified algorithm for mode adjustment on standard GSM/3G systems. Neighboring patterns are in different classes that effectively partition the total number of patterns, as shown in the table in Fig. 7a. As shown in Figure 7b, if the transmission is in mode 11 and category 1 (12.2kbps), and the error estimate from the convolutional decoder indicates that the bit rate needs to be reduced, the mode command causes a mode in the same category to be shifted down 72, i.e. move to Mode 10 Category 1 (7.95kbps). If the BER is still too high, the command moves down a second time 74 to Mode 01 Category 1 (6.7kbps). The BER is still too high causing another shift down 76 to mode 00 category 1 (5.15kbps). If the BER is still too high, a final cross-class move 78 to mode 00 class 2 (4.74kbps) is performed. Thus, the entire range of available modes spans 4 command moves. Similarly, if the BER is reduced below a predetermined threshold, the command is moved up the category to increase the sending speed. For example, if the system is already in the lowest mode, i.e. in mode 00 category 2 (4.74kbps), and the BER is now improved, the command moves up 80 modes to mode 01 category 2 (5.9kbps), rather than going back to mode 00 category 1. A further improvement results in a move 82 to Mode 10 Category 2 (7.4kbps), and a still further improvement results in a move 84 to Mode 11 Category 2 (10.2kbps). If further improvement is available, the algorithm causes the mode to be moved 86 across classes, ie to mode 11 class 1 (12.2kbps) for the highest transmission speed. Improvement or deterioration of BER in the middle of the class range results in an up or down move within the class, as shown by moves 88 and 90 . The algorithm allows for cyclical movement based on patterns within a class, where movement to a class occurs only when the minimum or maximum sending capability in the class is reached.

按专利法规定,以上已经详细描述了本发明,本领域技术人员将意识到对这里公开的特定实施例的修改和替换。这些修改处于所附权利要求限定的本发明的范围和意图内。Having described the invention in detail above, those skilled in the art will recognize modifications and substitutions to the specific embodiments disclosed herein as required by the patent laws. Such modifications are within the scope and intent of the invention as defined in the appended claims.

Claims (10)

1. encode and error correction method for one kind, may further comprise the steps:
From a plurality of Legacy System frames, peel off header data with head and traffic channel TCH data block;
Collect and interior speech data of the essentially identical period of period of a plurality of time slots;
With the header data coding growth frame header that separates;
To described encoded speech data;
Speech data behind described long frame header and the coding is combined into long frame;
Described long frame is deconstructed into a plurality of equal segmentations; And
Described segmentation is sent as the form of TCH data with described Legacy System frame.
2. the method for claim 1, the coding of wherein said speech data comprise and adopt the AMR codec.
3. the method for claim 1, the wherein said header data that separates comprise that the gentle slow phase of Channel Identifier closes control data.
4. the method for claim 1, the coding of the wherein said header data that separates comprise uses fixedly convolution coder.
5. method as claimed in claim 2, wherein said AMR codec has various modes and described method also comprises the step of selecting codec mode based on channel quality.
6. method as claimed in claim 4, wherein said convolution coder are 1/2 convolution coders.
7. method as claimed in claim 5, the step of wherein said preference pattern be included in BER when worsening in a classification to Move Mode down, when the minimum classification limit, stride classification and move, when BER improves, in a classification, upwards move, and when the highest classification limit, stride classification and move.
8. encode and error correction system for one kind, comprising:
Be used for peeling off the device (20) of header data from a plurality of Legacy System frames with head and traffic channel TCH data block (12a, 12b, 12c, 12d);
Be used to collect with the essentially identical period of the period of a plurality of time slots in the device (18) of speech data;
The device (20) that is used for header data coding growth frame header that to separate;
Be used for device (24) to described encoded speech data;
Be used for the speech data behind described long frame header and the coding is combined into the device of long frame (32);
Be used for described long frame is deconstructed into the device of a plurality of equal segmentations (34a, 34b, 34c, 34d); And
Be used for described segmentation as the device of TCH data with the form transmission of described Legacy System frame.
9. coding as claimed in claim 8 and error correction system, the coding of wherein said speech data comprise employing AMR codec (18), and it has various modes, and described system also comprises the device that is used for selecting based on channel quality codec mode.
10. coding as claimed in claim 9 and error correction system, wherein at the described device that is used for selecting codec mode, also comprise and being used for when BERization in a classification to Move Mode (72,74,76) down, and when the minimum classification limit, stride the device that classification moves (78), in a classification, upwards move (80,82,84) with being used for when BER improves, and when the highest classification limit, stride the device that classification moves (86).
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