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     

High quality audio transform coding at 64 kbit/s

This paper presents a transform coding algorithm designed for audio coding at a bit rate of 64 kbit/s. It enables the transmission of a high quality stereo sound through the 2B channels of isdn. Although a complete system including framing, synchronization and error correction has been developed, only the bit rate compression algorithm is described here. A detailed analysis of the signal processing techniques such as the time/frequency transformation, the preecho reduction by adaptive filtering, the fast algorithm computations…, is provided. The use of psychoacoustical properties is also precisely reported. Finally, some subjective evaluation results and one real time implementation of the coder using the att dsp52c digital signal processor are presented.     

New error recovery strategy for 64 kbit/s video codecs

A new error recovery strategy for 64 kbit/s video codecs is described. Unlike existing strategies of error recovery in which a new frame is transmitted on the detection of a channel error, the new strategy uses previous frames which are known to be correct. Computer simulations of the technique are performed and subjective and quantitative results are obtained     

A 64 kbit/s Videophone codec with forward analysis and control

A new combination of coding methods for a 64 kbit/s transmission system for typical videophone situations is investigated. The codec structure is based on a standard hybrid discrete cosine transform (DCT) codec with temporal prediction. The picture is divided blockwise into changed and unchanged areas. One motion vector with subpel accuracy is computed and transmitted for each block of the changed area. For the forward analysis, the prediction error is calculated in the whole picture. Only the blocks with the highest prediction errors are updated by a DCT with a perception adaptive quantization. The number of DCT update blocks depends on the remaining bits after the transmission of the overhead information. The codec is controlled by a forward analysis of the prediction error and is not based on a buffer control. The spatial resolution of the source signal is reduced in two steps to prevent a codec overload caused by too much activity between two frames.     

A 64 kbit/s Integrated Visual Communication System--New Communication Medium for the ISDN

Seeing-while-talking has been a dream of mankind for over a 100 years since the invention of telephone. In the past, various trials were performed in spite of the difficulty in installing the network for the actual service. However, with the progress of ISDN and the advancement of digital signal processing technology, the environment has been changing rapidly. In this paper, an integrated visual communication system is described for the enhanced communication service at 64 kbits/s, the fundamental bit rate of ISDN. The roles of state-of-theart compression of the audio and video signals are discussed and an integrated transmission method based on the priority of the information content is proposed.     

Multimedia telehomecare system using standard TV set

Nowadays, there are a very large number of patients that need specific health support at home. The deployment of broadband communication networks is making feasible the provision of home care services with a proper quality of service. This paper presents a telehomecare multimedia platform that runs over integrated services digital network and internet protocol using videoconferencing standards H.320 and H.323, and standard TV set for patient interaction. This platform allows online remote monitoring: ECG, heart sound, blood pressure. Usability, affordability, and interoperability were considered for the design and development of its hardware and software components. A first evaluation of technical and usability aspects were carried forward with 52 patients of a private clinic and 10 students in the University. Results show a high rate (mean = 4.33, standard deviation--SD = 1.63 in a five-points Likert scale) in the global perception of users on the quality of images, voice, and feeling of virtual presence.     

Low delay coder (< 25 ms) of wideband audio (20 Hz-15 kHz) scalable from 64 to 32 kbit/s

A low delay coder for speech and music signals sampled at 32kHz is described. Its algorithmic delay does not exceed 25 ms which enables audioconferencing applications without echo cancellation. Its bit rate is scalable between 64 and 32 kbit/s by steps of 8 kbit/s. The transmitter issues the binary code at 64 kbit/s with lower bit rate codes embedded in it. The receiver may operate at lower bit rates with gradual loss of quality. The proposed coder is based on a mixed scheme : the adopted solution contains elements from the CELP speech coder and frequency domain music coders. The perceptual signal is obtained in the time domain, then transformed to the frequency domain where bit allocation is calculated and transform coefficients are quantized. A first solution based on the dft is discussed, then a second solution based on a mdct with small overlap is applied. The quantization of these coefficients is done in the following way. First, a prediction of the whole spectrum is applied. Then, a mean- removed gain- shape split vq is used for amplitude spectrum quantization and a hierarchical 2- dimensional vq is used for phase spectrum quantization with amplitude correction. At the phase quantization stage, each codeword describing the selected vector index is split into parts corresponding to different bit rates. Due to the hierarchical codebook structure, truncated indices may be used, without much affecting the signal quality. Simulation results are presented and the robustness of the proposed coder is examined.     

APC-AB codec modules operating at 16 and 8 kbit/s

High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme. An optimized APC-AB algorithm is studied that reduces processing complexity while maintaining speech quality. The coding algorithm is implemented in two digital signal processors (DSPs). The DSP chips, a framing LSI circuit, a PCM codec, and some peripheral ICs are integrated in each of two compact packages, i.e. codec modules, operating at 16 or 8 kb/s. The codec module size is as small as 80 mm×50 mm×12 mm, and its typical power consumption is 500 mW using 2-μm CMOS LSI technology. At 16 kb/s this APC-AB codec achieves high speech quality, close to that of a 7-bit μ-law PCM. The codec modules are expected to be used for various applications such as customer premises multiplexers for digital leased lines, digital mobile radio, and stored-and-forward-message systems (voice-mail systems)     

基于多带的2.4 kbit/s波形内插算法

由于传统特征波形内插语音编码算法对特征波形相位信息的忽略,以及对特征波形的整体对齐,往往造成语音高频谐波分量丢失,从而导致语音的噪声感。为了提高合成语音的质量,该文引入语音多带清浊音标志,并以此为依据对波形内插编码模型中的慢渐变波形和快渐变波形的相位谱进行估计,在语音合成时则对特征波形采取部分对齐的方法,最后提出了一种基于多带的2.4 kbit/s特征波形内插算法。与传统算法相比,新算法明显提高了语音的清晰度。与标准2.4 kbit/sMELP算法相比,该算法合成语音质量亦略显优势。     

一种新的2kbit/s波形内插语音编码器

基于传统的波形内插语音编码模型,提出了一种新的2kbit/s语音编码方案。该方案在编码端去除了传统方法中复杂的对齐运算,在译码端用三次B样条插值取代传统的线性插值。慢渐变波形只量化低频分量,而快渐变波形用正交多项式拟合,并采用合成-分析技术对其进行矢量量化。DRT测试结果表明,该2kbit/s语音编码方法能获得高可懂度的重建语音。     

基于小波变换的2.4kbit/s波形内插语音编码算法

基于双正交小波滤波器组对波形内插编码中提取的特征波进行多级分解与重构，提出了一种基于小波变换（WT）的2．4kbit／s特征波形内插（CWI）语音编码算法。编码端去除了特征波对齐运算，并对幅度谱进行多级分解，相位谱不传输，鉴于小波变换对信号的压缩特性，仅传输对人耳感知起主要贡献的最后一级特征波幅度谱；解码端对各尺度空间采用单独重建的方法，相位信息在重构的末级与幅度谱结合，并由浊音度标志选择固定或随机相位。此外，根据语音信号的时变特性，由基于子帧的浊音度标志选择需要传输的幅度谱及量化模式。主观R-A／B测试表明，这种基于小波变换的2．4kbit／s编码算法的合成语音质量明显优于标准的2．4kbit／s的MELP编码器及FS1016的4．8kbit／sCELP编码器，亦优于3．8kbit／s的传统CWI编码框架下的合成语音效果。     

Vector-quantised transform coder for speech coding at 9-6 kbit/s and below

The transform approach to speech coding has been established for some time, and has been shown to be very efficient in controlling the bit allocation and the shape of the noise spectrum. Various transform coders have been reported which produce high-quality digital speech at around 16 kbit/s. Although these coders can maintain good quality down to about 9.6 kbit/s, they perform poorly at lower bit rates. Here we discuss how vector quantisation (VQ) can be used to improve the quality of transform coders. We describe one specific design of vector-quantised transform coder (VQTC) which follows on from earlier work, and which is capable of producing good-quality speech at as low as 4.8 kbit/s.     

Digital Transmission of Commentary-Grade (7 KHz) Audio at 56 or 64 Kbits/s

A two-band coding system has been constructed for the purpose of providing commentary grade (7 kHz bandwidth) speech or music transmission at 56 or 64 kbits/s. The lower band, 0-to-3650 Hz, is coded with 4 bit ADPCM and the upper band, 3600-to-6800 Hz, is coded with 3 bit or 4 bit/sample APCM. The quality of the coded signal makes the method useful for news and sports broadcasts, and possibly for AM remote music broadcasting. The audio sounds better than that produced by two conventional alternatives: 3200 Hz bandwidth with 8-bit/sample coding and 7000 Hz bandwidth with a single 4-bit/sample coder. The sample may be used in any place with access to a 56 kbit/s Dataphone Digital Service port or to other 56 or 64 kbit/s lines. The power consumption is approximately 12 W in the present form; it could be reduced by a factor of at least two by hardware optimization.     

Impact of frequency-selective fading on digital land mobile radio communication at transmission rates of several hundred kbit/s

Based on measurements reported from New York City and the city of Berne, Switzerland, and its outskirts, computer simulations of the mobile radio channel transfer functions were performed. It is demonstrated that the impulse response may suffer considerably from large variations of the channel characteristics. It is concluded that powerful means of signal processing are necessary to safeguard digital land mobile radio communication at transmission rates of several hundred kbit/s.     

A class of analysis-by-synthesis predictive coders for high qualityspeech coding at rates between 4.8 and 16 kbit/s

The general structure of this class of coders is reviewed, and the particulars of its members are discussed. The different analysis procedures are described, and the contributions of the various coder parameters to the performance of the coder are examined. Quantization procedures for each transmitted parameter are given along with examples of bit allocations. The speech quality produced by these coders is high at 16 kb/s and good at 8 kb/s, but only fair at 4.8 kb/s. The use of postprocessing techniques changes the performance at lower rates, but more research is needed to further improve the coders     

Design Considerations for a 144 kbit/s Digital Transmission Unit for the Local Telephone Network

In this paper a digital transmission unit for the forthcoming integrated services digital network (ISDN) is considered. Basic elements of such a unit are an adaptive echo canceller (EC) and an adaptive decision feedback equalizer (DFE). Each of these filters can be realized by a transversal filter structure or by a lookup table structure. An analysis and comparison of the behavior of these two structures is given, The combination of an EC and a DFE can also be realized by a single lookup table. It is shown that irrespective of the particular realization three different convergence phases can be distinguished, and that the overall convergence time is considerably longer then would be expected from the usual simple theory that ignores error propagation in the DFE. Two variants of a hardware realization of such a digital transmission unit are presented. They provide 144 kbit/s full-duplex transmission on the symmetrical cables of the existing local telephone network. A small transmission bandwidth is obtained by applying NRZ signaling. In the first circuit the EC and DFE are implemented by individual lookup tables, while in the second a combined table is used. Both circuits are to a large extent, implemented digitally. Due to the relatively low sampling frequency and the acceptably low power consumption they are very well suited for large scale integration with present day technology.     

40 Gbit/s single-channel transmission over standard singlemodefibre using distributed Raman amplification

A 40 Gbit/s single-channel transmission experiment over standard singlemode fibre has been conducted using distributed Raman amplification. Distributed Raman amplification reduced the effect of pulse interactions and improved the transmission performance     

20 Gbit/s soliton transmission using alternating dispersionarrangement in standard fibre systems

An alternating dispersion arrangement is proposed for soliton systems with dispersion compensation to improve soliton stabilisation. A transmission experiment at 20 Gbit/s over 2600 km long singlemode fibre was successfully demonstrated by employing this novel dispersion arrangement     

64-Gb/s highly reliable network switch (RHiNET-2/SW) using paralleloptical interconnection

RHiNET-2/SW is a network switch that enables high-performance optical network based parallel computing system in a distributed environment. The switch used in such a computing system must provide high-speed, low-latency packet switching with high reliability. Our switch allows high-speed 8-Gb/s/port optical data transmission over a distance of up to 100 m, and the aggregate throughput is 64 Gb/s. In RHiNET-2/SW, eight pairs of 800-Mb/s×12-channel optical interconnection modules and a one-chip CMOS ASIC switch LSI (a 784-pin BGA package) are mounted on a single compact board. To enable high-performance parallel computing, this switch must provide high-speed, highly reliable node-to-node data transmission. To evaluate the reliability of the switch, we measured the bit error rate (BER) and skew between the data channels. The BER of the signal transmission through one I/O port was better than 10^-11 at a data rate of 800 Mb/s ×10 b with a large timing-budget margin (870 ps) and skew of less than 140 ps. This shows that RHiNET-2/SW can provide high-throughput, highly reliable optical data transmission between the nodes of a network-based parallel computing system