Speech coding in mobile radio communications

Speech coding, the efficient representation of speech in digital form, is one of the key technologies in current and evolving digital cellular and wireless voice communications offerings. The speech coders in existing standards exhibit a level of sophistication and performance unimaginable just 15 years ago. We outline the characteristics of the mobile communications problem with respect to speech coders and point out the principal issues in speech coder design for these applications. Speech coding methods in existing mobile communications standards are described and contrasted. The limitations imposed by the wireless channel and by background impairments are discussed, and approaches to addressing their resulting effects are presented. Suggestions for future research in speech coding for the mobile communications problem are outlined     

A Comparison of Digital Speech Coding Methods for Mobile Radio Systems

Subjective quality measurements on three digital speech coders, simulated with mobile radio channel transmission, were performed using the "mean opinion score (MOS)" method. The three speech coding methods tested were: continuously variable slope deltamodulation (CVSD) coding, adaptive predictive coding (APC), and residually excited linear predictive (RELP) coding. Several versions of each coder, with transmission rates in the range of 7.3 to 16.1 kbits/s, were simulated. Five different channel conditions, including three derived from land mobile radio field experiments, were applied to the speech coders' encoded output to study the effects. The results show that of the three coders, the CVSD coder is the most robust to channel errors, but produces reconstructed output speech of unacceptable quality. The 14.4 kbit/s RELP coder produces relatively good Output speech quality, exhibits a mild degree of robustness to mobile radio channel errors, and is slightly less complex than the APC coder. Of the three digital speech coders tested, the RELP coder appears the most suitable for use with land mobile radio. However none of the three coders was able to produce speech of telephone toll quality in a mobile radio environment.     

自适应多码率语音编码流的可靠传输

自适应多码率语音编码已入选为第三代移动通信系统的语音压缩编码方案。本文提出了一种适合压缩语音传输的联合信源信道编码方法并对其性能进行了统计比较。利用压缩语音比特流中的固用冗余的信道译码算法是本文的研究重点。仿真结果表明利用信源冗余信息的信道译码器可以获得较大的编码增益。本文所用的信道编码方案为适合语音传输的卷积码。     

Advances in coding and compression

The significant activity that has taken place in China on source coding is reviewed. This includes three of the most important source coding techniques used for speech compression-waveform coders, vocoders, and hybrid coders-and a number of techniques used to implement image compression. Two other areas of active communications research, channel coding and cryptography, are discussed     

Adaptive prediction in speech differential encoding systems

The design of speech coders that produce high-quality highly intelligible speech at 6 to 16 kb/s while retaining robustness to background and transmission impairments is an area of current research interest. Differential encoding structures employing adaptive quantization and adaptive prediction constitute one of the most promising approaches to achieving these design objectives. This paper focuses on the design and analysis of adaptive predictors for differential encoders. Several differential encoding systems, including adaptive predictive coding, differential pulse-code modulation, noise feedback coding, direct feedback coding, and prediction error coding, are described and related. Adaptive quantizers are briefly discussed and quantitative and qualitative indicators of speech coder performance are defined. The channel model, the speech model, and the research problem statements used in the design of differential encoders and adaptive predictors are presented. The nomenclature and theory of forward and backward adaptive prediction are developed, and several new backward adaptive algorithms based on various assumptions are presented. A detailed survey of theoretical and simulation results on adaptive prediction for speech differential encoders is given, and the effects of background and transmission impairments on these systems are discussed, Finally, the impact of adaptive predictors on rate distortion theory motivated coders is indicated. Numerous areas for future research are highlighted.     

一种自适应分组丢失率的多描述正弦语音编码器

多描述（MD）语音编码器可以在不可靠的信道如internet上稳定地传输语音信号。然而,当前的MD语音编码器一般对不同的分组丢失率采用固定的多描述结构,不能很好地适应实际网络环境中分组丢失率的实时变化。该文提出一种自适应多描述正弦编码器(AMDSC),可根据分组丢失率的大小在两个描述间动态地分配冗余,从而使最终的重建失真最小。仿真结果表明,AMDSC的重建语音质量相对于其他固定结构的MD编码器有明显改善。     

Arithmetic coding-based continuous error detection for efficientARQ-based image transmission

Block cyclic redundancy check (CRC) codes are typically used to perform error detection in automatic repeat request (ARQ) protocols for data communications. Although efficient, CRCs can detect errors only after an entire block of data has been received and processed. We propose a new “continuous” error detection scheme using arithmetic coding that provides a novel tradeoff between the amount of added redundancy and the amount of time needed to detect an error once it occurs. This method of error detection, first introduced by Bell, Witten, and Cleary (1990), is achieved through the use of an arithmetic codec, and has the attractive feature that it can be combined physically with arithmetic source coding, which is widely used in state of-the-art image coders. We analytically optimize the tradeoff between added redundancy and error-detection time, achieving significant gains in bit rate throughput over conventional ARQ schemes for binary symmetric channel models for all probabilities of error     

Medical image transmission over a GSM cellular system

This authors report on an investigation of possible source/channel coders for medical image transmission over a GSM cellular link. Two source coders (JPEG and EZW) and two channel coders (convolutional and turbo codes) are used. Analysis of the results indicates that EZW with convolutional coding gives the best performance, but some kind of automatic repeat on request (ARQ) is also needed     

New directions in subband coding

Two very different subband coders are described. The first is a modified dynamic bit-allocation-subband coder (D-SBC) designed for variable rate coding situations and easily adaptable to noisy channel environments. It can operate at rates as low as 12 kb/s and still give good quality speech. The second coder is a 16-kb/s waveform coder, based on a combination of subband coding and vector quantization (VQ-SBC). The key feature of this coder is its short coding delay, which makes it suitable for real-time communication networks. The speech quality of both coders has been enhanced by adaptive postfiltering. The coders have been implemented on a single AT&T DSP32 signal processor     

一种面向保密通信的改进型MBE语音编码算法

随着军事通信的应用需求迅速扩展,如何有效地在信源端对语音信号进一步压缩。并且在复杂信道条件下实现高质量的低速率语音编码技术是一个重要研究方向。以MBE语音编码模型为基础,提出了一种改良算法,即在编码端利用信源冗余度,将对语音合成质量影响较大的参数进行检纠错保护,并在解码端采用谐波增强以改善终端语音合成质量。测试数据表明,在1％-3％的信道误码条件下,PESQ评分平均提高了近14％。     

Pseudo-analog speech transmission in mobile radio communicationsystems

A low-complexity pseudo-analog speech transmission scheme is proposed for portable communications. It uses a speech coder based on adaptive differential pulse code modulation (ADPCM) in combination with a multilevel digital modulation technique such as M-ary DPSK or M-ary FSK and features low quantization noise, bandwidth efficiency, and robustness to transmission errors. A nonsymmetric M -ary DPSK scheme called skewed M-ary DPSK is proposed to enhance the noisy channel performance. Comparison to conventional analog FM and a digital speech transmission scheme using adaptive predictive coding and forward error correction (FEC) based on convolutional coding shows that the pseudo-analog system has the best objective signal-to-noise ratio performance under most channel conditions. Informal subjective evaluations rate the digital system superior to the pseudo-analog scheme for bad channels and conversely for good channels. It is concluded that the pseudo-analog system can be designed with low delay and high speech quality for good channels with high spectral efficiency     

Joint source-channel (de-)coding for mobile communications

Real world source coding algorithms usually leave a certain amount of redundancy within the coded bit stream. Shannon (1948) already mentioned that this redundancy can be exploited at the receiver side to achieve a higher robustness against channel errors. We show how joint source-channel decoding can be performed in a way that is applicable to any mobile communication system standard. Considerable gains in terms of bit error rate or signal-to-noise ratio (SNR) are possible dependent on the amount of redundancy. However, an even better performance can be achieved by changing also the transmitter sided source and channel encoders. We propose an encoding concept employing low-dimensional quantization. Keeping the gross bit rate as well as the clean channel quality the same, it decreases the complexity of the source encoder and the decoder significantly. Finally, we give an application of our methods to spectral coefficient coding in speech transmission over a Rayleigh fading channel resulting in channel SNR gains of about 2 dB as compared to state-of-the-art (de-)coding and bad frame handling methods     

Digital audio coding for visual communications

Current and future visual communications for applications such as broadcasting videotelephony, video- and audiographic-conferencing, and interactive multimedia services assume a substantial audio component. Even text, graphics, fax, still images, email documents, etc. will gain from voice annotation and audio clips. A wide range of speech, wideband speech, and wideband audio coders is available for such applications. In the context of audiovisual communications, the quality of telephone-bandwidth speech is acceptable for some videotelephony and videoconferencing services. Higher bandwidths (wideband speech) may be necessary to improve the intelligibility and naturalness of speech. High quality audio coding including multichannel audio will be necessary in advanced digital TV and multimedia services. This paper explains basic approaches to speech, wideband speech, and audio bit rate compressions in audiovisual communications. These signal classes differ in bandwidth, dynamic range, and in listener expectation of offered quality. It will become obvious that the use of our knowledge of auditory perception helps minimizing perception of coding artifacts and leads to efficient low bit rate coding algorithms which can achieve substantially more compression than was thought possible only a few years ago. The paper concentrates on worldwide source coding standards beneficial for consumers, service providers, and manufacturers     

低比特率语声编码器的新发展

ITU-T建议G.729、G.729 AnnexA和G.723.1是国际电信联盟(ITU)最新颁布的3种适用于多媒体通信的低比特率线性预测语声编码器标准。文章介绍了语声编码器的比特率、复杂度、延迟和音质等性能指标的含义,并通过比较3种标准的新型声码器在算法和性能指标上的异同点,讨论了它们在多媒体通信中的不同应用。     

Packet splitting and adaptive modulation based on CSI of time domain for decode‐and‐forward cooperative OFDM systems in different channel model

Cooperative communications obtain the transmission and channel diversity gains by using the relay node. However, since cooperative communications transmit the redundancy signal to obtain the transmission diversity gain, the transmission rate is degraded. Moreover, since cooperative communications add the interference in the relay node, the diversity gain is also degraded. The packet splitting has been proposed based on the channel state information of the time domain to obtain the good system performance without the redundancy signal. Moreover, the adaptive modulation has been proposed to improve the transmission rate. In this paper, we propose the combination method with the packet splitting and the adaptive modulation based on the channel state information of the time domain to improve the bit error rate and throughput performances for decode‐and‐forward cooperative orthogonal frequency division multiplexing systems in the different channel model. From the computer simulation results, we determine the optimum weight and threshold for the proposed method. Moreover, the proposed method shows the good bit error rate and throughput performances.     

A joint source/channel coder with block constraints

The assumptions made about the source during source coder design result in a residual redundancy at the output of the source coder. This redundancy can be utilized for error protection without any additional channel coding. Joint source/channel coders obtained using this idea via maximum a posteriori probability decoders tend to fail at low probability of error. In this paper, we propose a modification of the standard approach which provides protection at low error rates as well     

Sliding window criterion codes and concatenation scheme formitigating timing-jitter-induced errors in WDM fiber transmissions

We propose a concatenated coding scheme, which effectively reduces bit errors induced by soliton-soliton collisions (SSC) in wavelength division multiplexing (WDM) soliton transmission systems. A block line coding scheme, the sliding window criterion (SWC) code, is developed based on the nature of SSC-induced timing jitter in soliton communications. We show, by simplified collision model simulations, that the SWC code alone can decrease the SSC-induced timing jitter and, by concatenation to a Reed-Solomon (RS) code, improve both the bit rate and the channel spacing capacity in WDM systems. We compare the performance of our scheme both analytically and by simulations with those of various RS codes and concatenated RS-convolutional code used in optical fiber transmission systems, and show that high redundancy (overhead) does not always give better code performance. Finally, by using full simulations, we show that the SWC code is an effective and promising technique for dispersion-managed fiber WDM systems