A packet-based CAPDM speech coder for PCN applications

In this paper, we present a median-rate speech coder, the controlled adaptive prediction delta modulation coder (CAPDM), which operates at 16 kb/s with good speech quality and low algorithm complexity. The coder is dedicated to personal communication network (PCN) applications and transmits speech samples on the basis of packets. It combines the features of a one-step looking forward decision, syllabic companding, instantaneous companding, and adaptive prediction. In addition to the use of a short-term prediction filter, CAPDM also exploits the pitch property to predict speech waveform explicitly. With the aid of a pitch prediction filter, the performance of a CAPDM codec improves about 3 dB in segmental signal-to-noise ratio (SEGSNR). The average SEGSNR of CAPDM.FF is about 21 dB, which is 7 dB over traditional CVSD at 16 kb/s. We also utilize an adaptive postfilter (APF) to enhance the perceptual quality of the decoded speech. The mean opinion score (MOS) listening test of CAPDM.FF with APF shows that its average score achieves 4.19, which is as good as G.728 16-kb/s LD-CELP and is comparable with CCITT G.721 32-kb/s ADPCM. The complexity of CAPDM.FF is evaluated to be 8 MIPS, which is much lower than that of LD-CELP and could be further reduced by adopting a smaller correlation window for pitch detection. To solve the problem of packet loss, we developed a packet-based waveform substitution method by reinitializing the codec parameters at the beginning of each packet. The simulation results show that CAPDM.FF could tolerate 5% of packet loss and still keep an SEGSNR at 10 dB and an MOS at about 3.0     

64 kbit/s audio signal transmission approaches using 32 kbit/sADPCM channel banks

Two simple 64-kb/s wideband coding approaches using 32-kb/s ADPCM (adaptive digital pulse-code modulated) channel banks are proposed and compared to CCITT 64 kb/s ADPCM, which is being recommended as CCITT G.722. These two, folding ADPCM and QMF ADPCM, are intended to pave the way for smooth transition from conventional 4-kHz band telephone systems to 7-kHz wideband systems in private networks. The first approach, supporting the high-quality audio program transmission, requires only samplers and multiplexers at the input and output ports of the channel banks. In the second approach, samplers and multiplexers are replaced by quadrature mirror filters in order to increase coding quality. Performance test results for audio signal transmission show that these simplified approaches provide an inexpensive way to introduce wideband communication systems     

语音数字信号压缩技术在TES系统中的应用

介绍ADPCM标准、RLPC编码原理,编、解码器方框图及工作过程。民航卫星通信网TES系统为节省卫星转发器频率资源对传输的语音信号进行压缩处理,其信道单元基带信号处理器对语音信号进行CCITT推荐的G.721-ADPCM编码和修斯公司专利技术开发的RLPC编码处理,将64 kb/s语音数字信号压缩至32 kb/s,16 kb/s,9.6 kb/s传输,实现语音质量满足一般通信要求的低速率语音信号传输。     

Overview and performance of CCITT/ANSI embedded ADPCM algorithms

Embedded adaptive differential pulse coded modulation (ADPCM) algorithms quantize the differences between the input signal and the estimated signal into core bits and enhancement bits. CCITT Recommendation G.727, which describes embedded ADPCM encoding algorithms with 5, 4, 3, and 2 core bits, is virtually identical to the corresponding ANSI standard T1.310. The main features of G.727 and T1.310 and performance results are presented. A formal subjective evaluation of the speech performance of embedded ADPCM algorithms indicates that a midrise quantizer provides better voice transmission performance than its midtread counterpart when two core bits are used. The subjective data also show that the performance of the 40-kb/s midrise ADPCM algorithm with two feedback bits is indistinguishable from that of 64-kb/s pulse code modulation (PCM) for up to four tandem encodings. Embedded algorithms are therefore recommended for flexible congestion control of integrated traffic in multinode networks     

基于改进RBF神经网络的ADPCM语音编码算法研究

以G．721ADPCM语音编码算法为研究对象，在语音编码的预测中引入神经网络模型来克服传统线性滤波方法中存在的不足，研究了基于RBF神经网络的ADPCM语音编码系统的结构。通过k均值聚类算法来确定RBF神经网络的中心和宽度，用最小二乘法确定RBF网络权值的方法改进了ADPCM语音编码算法。实验证明．其平均信噪比较原ADPCM编码算法有1-2dB的提高。     

ADPCM语音解码合成输出系统的设计

文章介绍了自适应差分脉冲编码调制(ADPCM)技术的编解码和脉冲宽度调制(PWM)技术的基本原理,研究在现场可编程门阵列(FPGA)上通过有限状态机方式实现ADPCM语音解码算法,利用PWM技术将解码后的数字语音信号转化为PWM波,以此直接驱动喇叭发出声音,输出的合成语音质量良好.     

60路32kb／s ADPCM转换器的大规模集成电路设计

３２ｋｂ／ｓ ＡＤＰＣＭ转换设备的研制在数字通信领域具有显著的社会与经济效益。本文的工作是在用中小规模集成块构成了６０路转换编译码系统样机的基础上，进行了大规模专用集成电路芯片的电路设计，拟用两片５０００门门阵列实现全系统集成，目前，线路级的设计已经完成，并且成功地通过了Ｄａｉｓｙ工作站的计算机逻辑模拟。     

基于TMS320F2812的多通道实时语音编解码

介绍了自适应多码率语音编解码算法及其基于TMS320F2812定点DSP芯片的实现方案。利用TMS320F2812芯片集成的多路ADC和PWM,对方案进行了多通道的扩展。分析了在DSP芯片上实现实时语音编解码及多通道扩展的关键技术。最后分析了此多通道实时语音编解码方案所需的存储空间和计算复杂度。     

ADPCM with a multiquantizer for speech coding

A speech coding algorithm with low complexity and a short processing delay is introduced. The proposed algorithm is ADPCM (adaptive digital pulse code modulation) with a multiquantizer (ADPCM-MQ). The input signal is processed in parallel by multiple ADPCM coders with different characteristics. Then the optimum ADPCM coder with minimum error power is dynamically selected for each frame. A 16-kb/s codec based on this algorithm has been implemented using two general-purpose digital signal processors (MB8764) with 8.3 ms of total processing delay. A segmental SNR of 19-21 dB was achieved at 16 kb/s; with postfiltering the segmental SNR was increased to 23-25 dB. Combined with the time domain compression scheme, the algorithm can be easily applied to 8-kb/s coding. It is also extensible to variable-rate coding     

Low-power and high-quality signal transmission baseband LSIC forpersonal communications

This paper presents a very low power consumption one-chip baseband large-scale integrated circuit (LSIC) for personal communication terminals. It comprises a π/4-shift QPSK modem, an adaptive differential pulse code modulation (ADPCM) codec, a time division multiple access time division duplex (TDMA-TDD) controller and a link access procedure for a digital cordless (LAPDC: Layer-2 protocol) controller. The developed LSIC meets all the specifications of the personal handy-phone system (PHS) standard. By employing a novel coherent demodulator and an ADPCM codec with a click noise suppressor, a higher quality voice transmission has been achieved in a fading environment. In addition, it has 61-kb/s data transmission capability to achieve wireless multimedia services based on PHS. Moreover, the novel circuit configurations of the modem, the ADPCM codec, the TDMA-TDD controller, and the LAPDC controller achieve significant power reduction of the baseband circuits (57.4 mW) of personal communication terminals. It enables very low power consumption wireless multimedia terminals to be achieved based on the PHS common air interface     

基于分数阶滤波器的ADPCM预测误差信号处理

ADPCM音频压缩算法在G．721,G．723和G．726等相关数字语音通信协议中得到广泛应用。针对ADPCM中普遍存在的预测误差信号镜频干扰和高频噪声的问题,提出了基于分数阶积分器的误差信号频谱压缩滤波迭代算法和基于分数阶积分器的逆系统信号恢复迭代算法,为进一步提高音频压缩编码品质提供了新的方法,理论分析和仿真结果表明所提出方法的可行性。     

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.     

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。     

基于CVSD编码的无线语音系统方案的设计

简要介绍了连续可变斜率增量(CVSD)调制的原理。与目前应用较为广泛的其它语音编码方式相比较,CVSD拥有更优的数字特性。着重介绍了由CML公司研制开发的基于CVSD的语音编码芯片———CMX649的特点及相关的应用方式。CMX649能够成功地应用在广泛的语音编码系统中,尤其适合无线语音系统应用。在此基础上,给出了一种实用性很强的低成本、低功耗无线语音系统的设计与应用方案。该方案可提供清晰可靠的语音传输,可广泛应用于农村地区,具有广阔的市场空间。     

A single-board video codec using video image signal processors

This paper discusses a newly developed single-board video codec using Video Image Signal Processors (VISPs). The codec has both a CCITT H.261 mode and a proprietary mode. Two VISPs, one for encoding and one for decoding, are used. The board size is 210 by 295 mm, the maximum frame rate is about 7.5 f/s, and the picture size is 180 by 144 pels.     

Sequential Adaptive Predictors for ADPCM Speech Encoders

The sequential gradient estimation predictor is compared in detail to the stochastic approximation predictor, and both are evaluated in an ADPCM codec. A switched predictor having two coefficients is then described for use in a DPCM-AQF codec. This predictor divides the range of the correlation coefficient of the speech signal into zones, and as the correlation coefficient changes zones, the predictor coefficients undergo a substantial modification. By this method the adaptation rate of the predictor is improved, particularly during transitions between unvoiced and voiced sounds.     

低功耗高速无线传输语音采集系统的设计

基于低功耗的无线传输协议802．15与低复杂度的语音编码压缩算法G．721（ADPCM）,开发出一套无线控制与传输的语音信号采集系统。实现了语音采集、无线传输、USB回传等功能。硬件方面主要采用射频调制芯片NRF24L01、微控制器芯片8051F206以及USB控制器芯片CY7C68013等。系统具有体积小、低功耗而语音信号质量与传输速率较高的特点。     

A monolithic ΣΔ A/D and D/A converter with filter forbroad-band speech coding

The authors present a 5-V-only 14-b, 16 ksamples/s linear codec suitable as the audio part of a CCITT G722 codec. The device uses second-order sigma-delta modulation for both analog/digital (A/D) and digital/analog (D/A) conversion at 2.048 Msamples/s. A time-continuous modulator with integrated antialias filtering is used at the A/D side, obviating the need for an external antialiasing filter. The digital filters for decimation and interpolation are implemented with both a custom digital signal processor (DSP) and specialized hardware. The device was realized with 74000 transistors on a 31-mm² die in a 3-μm SACMOS technology. A dynamic range of more than 80 dB and a passband ripple of 0.3 dB were attained with A/D and D/A paths in cascade