Performance Analysis of PCM/ADPCM Transcoding Systems

In this paper, the performances of PCM/ADPCM transcoding systems are analyzed. The coders studied are the 64 kbit/s PCM with μ-255 companding law and the 32 kbit/s ADPCM proposed by Cummiskey et al. The theoretically predicted performance agrees closely with the results of computer simulation for a wide range of input signal level. According to the results, the performance degradation resulting from the code conversion process appears to be minimal for the single tandem case. However, for the case of multiple tandem code conversions, the performance becomes significantly degraded as the number of tandem coders increases. The overall performance of the transcoding system depends largely on the performance of the coder that is inferior to the other coder being cascaded.     

A Microprocessor Log PCM/ADPCM Code Converter

A microprocessor has been used to translate between Log PCM and ADPCM (Adaptive Differential PCM) code forms. This system, in bridging the gap between simulation and prototyping, provides realtime speech processing with user interaction. Continuously coded speech can he subjectively evaluated while switching the values of code word length, step size, or predictor coefficients. Translations of additional code forms such as Δ-Mod, NIC, or Tree Codes could easily be implemented with the micro-codable system. The processor is configured as a stand-alone device competitive with special purpose hardware in size, speed, and cost.     

The Design of an ADPCM/TASI System for PCM Speech Compression

This paper describes the design of a digital speech interpolation (DSI) system called ADPCM/TASI for adaptive differential PCM with time assignment speech interpolation. This system is designed to compress the output of two T1 24-channel PCM carrier terminals into a 1.544 Mbit/s signal that can be transmitted over a single T1 carrier line. The design is based on a bit slice microprocessor structure. Alternative designs are also described.     

Shape-VQ-based lossless hybrid ADPCM/DCT coder

The discrete cosine transform (DCT) has been shown as an optimum encoder for sharp edges in an image (Andrew and Ogunbona, 1997). A conventional lossless coder employing differential pulse code modulation (DPCM) suffers from significant deficiencies in regions of discontinuity, because the simple model cannot capture the edge information. This problem can be partially solved by partitioning the image into blocks that are supposedly statistically stationary. A hybrid lossless adaptive DPCM (ADPCM)/DCT coder is presented, in which the edge blocks are encoded with DCT, and ADPCM is used for the non-edge blocks. The proposed scheme divides each input image into small blocks and classifies them, using shape vector quantisation (VQ), as either edge or smooth. The edge blocks are further vector quantised, and the side information of the coefficient matrix is saved through the shape-VQ index. Evaluation of the compression performance of the proposed method reveals its superiority over other lossless coders     

Quantization Noise in ADPCM Systems

The system considered here consists of a differential pulse-code modulator (DPCM) in which the quantizer is replaced by an adaptive quantizer. Adaptation is accomplished by adjusting the stepsize at every sampling instant, depending upon the magnitude of the quantization error. Quantizing noise in adaptive DPCM (ADPCM) systems falls into three categories: granularity, slope overload, and quantizer saturation. Granular noise occurs because only a finite number of levels are available to represent the analog input signal during the encoding process. Slope overload happens when the slope of the input signal increases faster than the adaptive system can follow. Quantizer saturation exists because nonoptimal decisions on step-size adjustments can lead to situations in which quantizer overload occurs without slope overload. The result is an error larger than a purely granular noise analysis would suggest. Equations for the three types of quantizing noise in ADPCM systems are derived, and computer simulations are perforated. For flatand RC-filtered Gaussian input signals, oversampled at various rates, the simulation results agree well with theoretical predictions. Comparisons indicate that the ADPCM can perform better than the best analogous nonadaptive system in terms of signal-to-quantizing-noise ratio. Furthermore, the optimal operating point for the ADPCM is much less sensitive to changes in input signal parameters and system component values than in nonadaptive systems.     

Comparative study of 24 kb/s ADPCM algorithms

Thepaper is devoted to comparison of four algorithms of 24 kb/sADPCM, standardADPCM (ADPCM-1), and three new modified algorithms (ADPCM-2, ADPCM-3, ADPCM-4). The purpose of the modified algorithms is to reduce the nonlinear distortion introduced by ADPCM when high data rate signal passes through it. The performances of the four algorithms are researched using QAM signal at data rate of 9.6kb/s. The simulation results show that the performance of ADPCM-4 is better than that of ADPCM-3, and the performance of ADPCM-3 is better than that of ADPCM-2, and the performance of ADPCM-2 is better than that of ADPCM-1.     

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.     

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     

Enhancement in the 32-kbit/s ADPCM algorithm for FSK voice-banddata signals

Due to the feedback architecture of the standard CCITT and ANSI 32-kb/s ADPCM (adaptive digital pulse-code modulated) coders, large transitional errors occur when characteristics of the input signal suddenly change. This particularly holds for FSK modems that transmit a single-frequency signal between actual characters. A mechanism that identifies these transitions and reduces errors is presented     

32kb／sADPCM中高速乘法器的设计

本文介绍了６０路３２ｋｂ／ｓＡＤＰＣＭ专用芯片中的高速乘法器的逻辑设计和提高运算速度的方法。通过优化设计，该乘法器运算速度高，电路简单，对芯片制造工艺要求不高。     

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。     

Effect of Carrier Mistracking and Phase Jitter in a QPRS Receiver

A straightforward procedure is described for estimating the effect of carrier mistracking and phase jitter on the error rate of a digital microwave radio receiver using a Quadrature Partial Response Scheme (QPRS) for modulation. Graphical results presented can be readily used to gain insight into the performance of a QPRS receiver.     

Quantization error and step-size distributions in ADPCM

An adaptive differential pulse code modulator (ADPCM) with a finite number of possible step-sizes and a leaky integrator in the feedback loop is considered. A zero-mean unit-ariance first-order Markov sequence is chosen as the input to thc system, and the leak parameter of the ADPCM accumulator is made equal to the intersample correlation of the input sequence. Using this fundamental structure, a method is presented for computing the exact joint probability distribution function of quantization error and step-size in ADPCM. From the joint distribution, marginal quantization error, and step-size distributions are obtained for Gauss-Markov, exponential-Markov, and uniform-Markov input signals. Empirical distributions obtained from simulations agree very well with their theoretical counterparts.     

The Power Spectrum of PCM/FSK-AM/PM

This paper derives the power spectrum of a modulating scheme under consideration by NASA for commanding scientific satellites. In this scheme the commands are frequency-shift keyed onto a subcarrier, then the clock is amplitude modulated onto the resultant. This PCM/FSK-AM waveform is then phase modulated onto a prime carrier. The method used in the paper can be easily extended to derive the spectrums of alternate command waveforms, such as PCM/FSK-AM/ FM, PCM/PSK-AM/PM, etc.     

Sequentially Adaptive Backward Prediction in ADPCM Speech Coders

Several questions concerning the performance in ADPCM systems of sequentially adaptive backward predictors based on the adaptive gradient and Kalman-type algorithms are addressed. Using a Jayant-type adaptive quantizer, it is shown that for bit rates less than 16 kbits/s with second order predictors and for bit rates less 18.4 kbits/s with fourth order predictors, backward-adaptive predictors have a definite performance advantage over fixed-tap predictors, since the latter may cause system divergence. For higher bit rates, the adaptive gradient predictor offers no advantage over a second order fixed-tap predictor; however, the Kalman predictor produces a substantial performance increment over the fixed-tap predictor. It is also shown that the Kalman predictor maintains a significant advantage over the adaptive gradient predictor for all bit rates from 12.8 to 32 kbits/s. Finally, it is noted that the ADPCM system divergence that occurs for fixed, multiple-tap predictors and a Jayant quantizer is caused by predictor mismatch with the input signal coupled with the infinite quantizer memory. This problem can be corrected by a modification to the quantizer adaptation logic.     

多通道PCM/AMBE实时转换系统

AMBE是基于MBE模型的低比特率、高质量语音压缩编码,依据PCM／AMBE语音信号互相转换的意义及转换的基本原理,提供了一个多通道的PCM／AMBE实时转换硬件平台,给出了系统的性能参数以及应用前景。     

ADPCM语音编解码电路设计及FPGA实现

近年来，多媒体技术逐渐深入到人们的生活中。MP3播放器已经成为流行的便携式音频播放设备，由于MP3编码算法非常复杂，目前，一部分MP3播放器的录音功能主要基于ADPCM算法和DSP来实现。本文阐述了ADPCM语音编解码VLSI芯片的设计方法以及利用FPGA的硬件实现。     

基于FPGA的ADPCM语音编解码器设计实现

FPFA的可配置性很好适应了设计周期短、升级维护的要求。ADPCM算法有4:1压缩率,但抗噪音及实时性还需改进。16bit声音波形数据的压缩算法,编码率低,但实时性差。文中介绍该算法的原理,研究在FPGA上通过有限状态机方式实现该算法,设计了基于这种算法的编码器与解码器,改善了实时性,得到较好的编解码效果。     

ADPCM语音压缩编解码器的FPGA实现

通过对自适应差分脉冲编解码算法原理进行分析,讨论其在FPGA上的实现。实现过程中,充分利用了FPGA器件的优点,结合有限状态机的特点对算法进行了更好的简化和优化,提高了算法效率,使应用具有更好的可靠性和实时性。     

Backward Adaptive Lattice and Transversal Predictors in ADPCM

Four different backward adaptive predictors and a fixed predictor are compared for use in an adaptive differential pulse code modulation (ADPCM) system for coding speech at 16 kilobits/second (kbits/s). For noise-free channels, the four adaptive predictors, a least squares lattice, a least mean square lattice, a Kalman transversal form, and a gradient transversal form, all exceed the fixed predictor performance as well as the performance of a continuously variable slope delta (CVSD) modulation system. For bit error rates (BER's) of 10^-3or greater, the transversal predictor performance falls below that of the fixed predictor and CVSD; however, the lattice structures maintain their performance advantage. The least squares lattice predictor has the best objective and subjective performance for both noiseless and noisy channels. All systems perform poorly for a BER of 10^-2. To extend the performance of ADPCM with a least squares lattice predictor down to a BER of 10^-2, the sampling rate is reduced and a selective coding scheme is devised. The resulting ADPCM system maintains excellent performance through a BER of 10^-2and outperforms CVSD for noise-free and noisy channels. The dynamic range, tandeming performance, and behavior for noisy inputs for the ADPCM system and CVSD are investigated.