Entropy coding of ITU G.728 speech parameters for low bit rateaudiovisual communications

Entropy coding principles are applied to the 16 kbit/s ITU G.728 speech codec. It is shown that the average bit rate can be reduced to 14.5 kbit/s without a significant increase in the codec complexity. In very low bit rate audiovisual communication applications such as the videophone, the saved bits can be used to improve the output video quality     

Highly Sensitive Speech Detector and High-Speed Voiceband Data Discriminator in DSI-ADPCM Systems

This paper describes a highly sensitive speech detector and a high-speed voiceband data classifier capable of discriminating between speech and voiceband data of a 4.8 kbit/s 8-phase PSK and 4.8 kbit/s 8-point QAM, and a 9.6 kbit/s 16-point QAM as described in a CCITT recommendation. The presence of a speech signal is detected by analyzing short-time energies, zero-crossing rates, and sign bit sequences of the input signal. The proposed speech detector, with a short hangover time of 32 ms, is able to reduce the average talk spurt activity in an international satellite link to 36 percent. This detector can also classify the detected speech into narrow-band or wide-band spectrum sounds or a low power sound for a variable rate ADPCM encoding. Discrimination between speech and high-speed voiceband data is based on short-time energies, a zeros-crossing rate and linear prediction coefficients of an adaptive predictor. Classification among a 4.8 kbit/s 8-phase PSK and 8-point QAM, and a 9.6 kbit/s 16-point QAM can also be performed by an average prediction gain and a coefficient of variation of the short-term amplitude distribution of the input signal. Discrimination of voiceband data was performed successfully, and erroneous discrimination of talk spurt of telephone speech as voiceband data were, respectively, four times for two two-party conversations lasting 5 minutes in an international satellite link. This is equivalent to less than 0.09 percent of the conversation time.     

Algorithm of Adaptive Bit Allocation Wavelet Transform Audio Coding

ＡｌｇｏｒｉｔｈｍｏｆＡｄａｐｔｉｖｅＢｉｔＡｌｏｃａｔｉｏｎＷａｖｅｌｅｔＴｒａｎｓｆｏｒｍＡｕｄｉｏＣｏｄｉｎｇＭａＨｏｎｇｆｅｉＦａｎＣｈａｎｇｘｉｎＳｏｎｇＧｕｏｘｉａｎｇ（ＸｉｄｉａｎＵｎｉｖｅｒｓｉｔｙ，Ｘｉ’ａｎ７１...     

Integration of harmonic and analysis by synthesis coders

Hybrid coding of speech has been proposed to overcome the limitations of a single model in representing the wide variety of characteristics of human speech. A new hybrid coding algorithm, which combines harmonic and analysis by synthesis coding techniques, is presented. To integrate the harmonic and analysis by synthesis coders, novel phase synchronisation and speech classification techniques are developed. The perceptual quality of the speech synthesised using the unquantised hybrid model is almost indistinguishable when compared with 128 kbit/s linear PCM. Two variable rate coders are developed based on the designed hybrid model, by quantising the parameters at different bit rates. Subjective listening tests show that the speech quality of the variable rate hybrid coders outperform the quality of 5.3 kbit/s and 6.3 kbit/s ITU G.723.1 coders, at maximum bit rates of 4 kbit/s and 6 kbit/s respectively.     

Subjective Evaluation of Several Efficient Speech Coders

The voice quality of several 9.6 - 32 kbit/s coders is determined with an extensive set of subjective listening tests. Single encodings of μ255 PCM, adaptive differential PCM (ADPCM), subband coding (SBC), vocoder-driven adaptive transform coding (ATC), adaptive predictive coding (APC), and time domain harmonic scaling combined with SBC are compared in an idealized situation, that is, no added impairments. It is shown that single encodings of modest complexity 32 kbit/s coders such as ADPCM and SBC and more complex 24 kbit/s coders such as vocoder-driven ATC and APC offer quality nearly equivalent to 64 kbit/s μ255 PCM. However, these conclusions are drawn in the absence of a realistic telephone network where tandem encodings, delay limitations, and nonvoice signals exist. Tandem encodings of 64 kbit/s μ255 PCM, 32 kbit/s ADPCM, 16 kbit/s SBC, and 16 kbit/s APC are also evaluated. These 32 kbit/s and 16 kbit/s coders offer degraded tandem performance as compared to 64 kbit/s PCM, with the exception of synchronous tandeming of 32 kbit/s ADPCM with 64 kbit/s PCM where several encodings are subjectively equivalent to a single encoding of 32 kbit/s ADPCM.     

基于DSP的多进制正交码扩频系统

本文介绍了一种基于ＤＳＰＴＭＳ３２０Ｃ２５的多进制正交码扩频系统，该系统可以对２．４ｋｂｉｔ／ｓ和１６ｋｂｉｔ／ｓ数据进行扩频。文中讨论了其实现的关键技术。     

基于VxWorks的嵌入式二次雷达航迹记录仪设计

为实现对二次雷达航迹数据的实时记录以及事后的回放分析,设计并研制了基于VxWorks实时系统的嵌入式记录仪,采用自定义缓冲队列的数据传输机制,充分考虑了任务之间的同步及互斥,并利用PC104板卡的硬件辅助时钟进行系统设计.实践证明,在航迹数据均值速率3.84 kbyte/s(96批/秒)(峰值速率为20.48kbyte/s即512批/秒)的情况下,能达到正常实况回放的效果.     

`Nearly instantaneous? digital compandor for transmitting six sound-programme signals in a 2.048 Mbit/s multiplex

A `near-instantaneous? digital compandor for the transmission of high-quality sound signals is described that reduces the bit rate from 416 kbit/s to about 322 kbit/s per channel without noticeable impairment of the sound quality. Hence six audio channels can be multiplexed to form a 2.048 Mbit/s stream including frame synchronisation and transmission error-protection facilities.     

Subband coding of image sequences at low bit rates

In this paper a low bit rate subband coding scheme for image sequences is described. Typically, the scheme is based on temporal DPCM in combination with an intraframe subband coder. In contrast to previous work, however, the subbands are divided into blocks onto which conditional replenishment is applied, while a bit allocation algorithm divides the bits among the blocks assigned for replenishment. A solution is given for the ‘dirty window’ effect by setting blocks to zero that were assigned to be replenished but received no bits. The effect of motion compensation and the extension to color images are discussed as well. Finally, several image sequence coding results are given for a bit rate of 300 kbit/s.     

深空通信Ka频段数传发射机基带电路单元实现

基于Xilinx FPGA电路的全数字化设计方案,研制完成适用于深空通信下行链路Ka频 段发射机中基带数据编码调制一体化电路单元。参照CCSDS（Consultative Committee for Space Data Systems）相关深空通信建议标准,电路单元实现了按码速率的变化灵活选择调 制方式的工作模式,利用外部控制指令,完成码速率16 bit/s～20 kbit/s、20～ 200 kbit/s、200 kbit/s～2 Mbit/s分段分别选择PCM/BPSK/PM、N RZ/BPSK和SRRC-QPSK数据调制方式 。在X频段的测试结果表明,BPSK和SRRC-QPSK幅度误差和相位不平衡分别小于3.1%和1.7° ,符合CCSDS关于深空通信的建议标准。电路单元满足深空通信工程应用需求。     

小波变换音频编码算法研究

任意能量有限信号都可以用紧支撑正交小波基展开或分解，这一点对研究快速高效音频编码算法是非常重要的。本文设计一种基于正交小波变换的高保真音频编码算法，该算法可以把速率为７０５．６ｋｂｉｔ／ｓ的高保真音频信号压缩到１９２ｋｂｉｔ／ｓ，１６０ｋｂｉｔ／ｓ，１２８ｋｂｉｔ／ｓ，９６ｋｂｉｔ／ｓ和６４ｋｂｉｔ／ｓ，并保持重构音频信号的高质量。     

基于小波变换和音质模型的音频编码算法研究

音频编码要解决的问题是以最小感知失真用低速率表达音频信号.本文设计了一种基于正交小波变换和音质模型的自适应比特分配音频编码算法,它可以将1411.2kbit/s的双声道立体声高保真音频信号压缩成低至32kbit/s的速率,并保持很好的音频质量.     

一种可变速率的LD—CELP编码的研究

设计了一种可变速率的低时延、码激励线性预测编码（ＬＤ－ＣＥＬＰ）的方案，它是通过修改码本来实现的。该方案工作在１１．２ｋｂｉｔ／ｓ。对其做了计算机仿真，并与１６ｋｂｉｔ／ｓ的ＬＤ－ＣＥＬＰ算法在信经（ＳＮＲ）、波形等方面进行了对比，仿真结果表明效果良好。     

Dual single-sideband modulation for wideband data transmission

A new modulation method for data transmission is described for use on groupband communication channels (60?108 kHz), which is compatible with the existing group- and supergroup-reference pilots. Dual single-sideband modulation permits a basic transmission rate of 48 kbit/s, which can be extended to 72 or 96 kbit/s by means of special multilevel coding.     

Implementation of a Multirate Speech Digitizer

In this paper, implementation of a compact and efficient multirate speech digitizer with variable transmission rates of 2.4, 4.8, 9.6, and 14.96 kbits/s is presented. The multirate algorithm has been made based on the residual-excited linear prediction (RELP) vocoder with a transmission rate of 9.6 kbits/s. The residual encoder employed in the RELP vocoder uses hybrid companding delta modulation (HCDM). This HCDM is also used as a 14.96 kbit/s coder. If the residual in the RELP system is down-sampled before encoding, a 4.8 kbit/s coder can be realized. If the residual encoder is not used, a 2.4 kbit/s linear predictive coder (LPC) can be realized by incorporating a pitch extractor. In the 4.8 and 9.6 kbit/s coders the pitch-implanted residual excitation method has been used to generate the excitation signal to the synthesis filter. The multirate speech digitizer algorithm has been implemented using 2900 series bit-slice microprocessors. The external memory is composed of 2K RAM's and 2K ROM's. The system design is a two-bus structure with a 204 ns cycle time. With efficient hardware and software design, the multirate speech digitizer requires almost the same hardware complexity as compared with the conventional 2.4 kblt/s LPC vocoder.     

宽带短波技术在测量船中的应用

针对现有岸船短波通信系统存在的数据传输速率低、多业务支持能力弱等问题,提出了一种宽带岸船短波通信系统。在应用数字调制技术和正交频分复用(OFDM)技术后,系统接收机灵敏度由-74 dBW降到了-91 dBW,数据传输率由9 kbit/s提升为50 kbit/s。对宽带岸船短波通信系统与其他岸船通信系统的多网融合潜力进行了分析,为新一代岸船通信网的建立提供了思路。     

量子密钥分发和经典光通信波分复用共纤传输研究

研究了量子密钥分发和经典光通信波分复用共纤传输的技术难点和可行性。基于系统重复频率 40 MHz的诱骗态相位编码BB84协议量子密钥分发设备,提出了3种量子信号与经典光信号的波分复用共纤传输方案：单纤双向CWDM共纤传输方案,复用1 550.12 nm波长量子信号、1 310 nm波长时钟信号以及正向1 590 nm波长100 Mbit/s速率光信号和反向1 610 nm波长100 Mbit/s速率光信号,光纤传输距离70 km下密钥成码率达到1.2 kbit/s;双纤双向CWDM共纤传输方案,复用1 550.12 nm波长量子信号、1 610 nm波长时钟信号以及1个波长的同向光信号在1 310 nm波长OOK光信号速率10 Gbit/s,光纤传输距离55 km下,密钥成码率达到1.58 kbit/s;双纤双向DWDM共纤传输方案,复用1 550.12 nm波长量子信号、1 610 nm波长时钟信号以及2个同向波长各自为1 551.72 nm和1 552.52 nm,并模拟100 Gbit/s相干光通信DP-QPSK信号接收功率,光纤传输距离70 km下,密钥成码率达到1.16 kbit/s。     

多媒体终端中声音和数据的集成传输

本文描述了采用包复用方式在固定带宽内集成传输声音和多媒体数据的多媒体终端通信系统，系统中的声音编码采用了静默检测技术，声音编码的速率可以根据信道的拥挤程度在３２ｋｂｉｔ／ｓ和１６ｋｂｉｔ／ｓ之间动态地变化。本文提出了一种利用增减静默抽样来同步声音编解码时钟的方法，本文还提出了利用数据队列的短时平均长度来判断信道繁忙程度的算法，在多媒体数据突发性强、数据量大时，该算法比利用声音或数据队列的瞬时长度判断更为准确。     

Performance of 16 kbit/s GMSK transmission with postdetection selection diversity in land mobile radio

Through laboratory simulation tests and field experiments in the Tokyo metropolitan area, 16 kbit/s Gaussian filtered minimum shift keying (GMSK) transmission performance has been experimentally clarified in the 920 MHz land mobile radio environment. The experimental results agree closely with theory, and they show that fast multipath fading severely degrades average bit error rate (BER) performance in GMSK transmission. However, a space diversity reception technique using a postdetection selection combining scheme is able to efficiently mitigate the fast multipath fading.     

Synchronous Tandem Algorithm for 32-kbit/s ADPCM

The synchronous tandem property of nonaccumulation of distortion in tandem-connected ADPCM coders with a 64 kbit/s PCM interface is discussed here. The synchronous tandem algorithm used to provide this property in the steady-state mode is described with a case-by-case analysis, so as to show how the synchronous tandem property is realized in an ADPCM coder. A 32 kbit/s ADPCM coder utilizing this algorithm has been standardized by the CCITT (International Telegraph and Telephone Consultative Committee). The synchronous tandem property of the 32 kbit/s ADPCM coder is of great interest in network applications, because the ADPCM coder appears likely to be introduced into digital networks built partially with existing 64 kbit/s PCM circuits.