Pulse Modulation

The problem of stability of an adaptive differential pulse code modulation (ADPCM) system is considered from two viewpoints. First, it is demonstrated that the step-size adaptation algorithm of a certain class of ADPCM's can be characterized by the system's response to step: function inputs. The degree of "hunting" or "oscillation" provides an indication of the deterministic stability of the algorithm, and algorithms are presented that minimize these effects. Secondly, stationary random processes with rational spectral densities are used to study stochastic stability of ADPCM. It is shown that for certain ADPCM's the time-averaged mean-absolute quantizing noise is bounded, that the joint distribution of the input and decoded output processes possesses a stationary distribution, that this joint distribution converges to the stationary distribution from an arbitrary starting point, and that the mean-absolute quantizing noise is finite under the stationary distribution.

In this paper, combination of the Pulse Width Modulation (PWM) technology and the Adaptive Differential Pulse Code Modulation (ADPCM) technology, design a digital synthetic voice chip. First of all, verilog hardware description language used to complete the design of decoding module, then the decoded digital voice signals transformed into PWM waves by using the PWM technology so as to drive the loudspeaker directly and sound. This type of voice synthetic output has such advantages as small power consumption, low cost and practical features.

New analytical formulas for quantization noise of two widely known adaptive delta modulation (ADM) systems, continuously variable slope delta modulation (CVSD) and Jayant's firstorder constant factor delta modulation (CFDM), have been obtained. These formulas are derived based upon the previously obtained results for linear delta modulation (LDM) and adaptive differential pulse code modulation (ADPCM). To verify the results, computer simulation has been done using band-limited Gaussian signals. Close agreements between the formulas and the simulation results have been obtained in the wide range of the input signal level.

Chapter 11 - Waveform Quantization and Compression

In this paper, we introduce an improved DPCM (Differential pulse Code Modulation) coding method that utilizes adaptive quantization and nonlinem (median type) predictors (Median Adaptive DPCM, MADPCM). DPCM is a promising coding method for television applications because of its simplicity and visual properties. The simulation results of the MADPCM coder obtained with the DVSR 100 digital video sequencer indicate that good picture quality can be achieved with a compression ratio of 2.7:1 (3 bits/pixel).

The myoelectric signal, obtained by either surface or needle electrodes, is used in many areas of clinical research and diagnosis. The conventional method of storing such information is in digitised form on a computer. However, the bandwidth of the signal and the required resolution result in large memory requirements. Adaptive differential pulse code modulation is investigated as a method of reducing the memory requirements for myoelectric data storage. In this scheme, a 12-bit sample is reduced to four bits, thus reducing the memory requirements by a factor of three. In reality, this compression ratio is closer to 4:1 owing to the fact that the widths of most memories are organised as multiples of eight bits.

Novel techniques are proposed to improve the adaptive differential pulse code modulation (ADPCM) voice signal transmission quality of time division multiple access-time division duplexing (TDMA-TDD) systems in fading environments. The proposed scheme employs an improved error detection technique with PCM signal differential detection and a cyclic redundancy code. Erroneous PCM bursts are replaced by error-free bursts. Hardware simulation results show that these techniques significantly improve ADPCM voice transmission quality in the fading environments typical of personal communication systems. The developed techniques enhance ADPCM voice transmission quality for cost effective personal communication systems.

In today's modern telephony network, VoIP is fast emerging as one of the main communication techniques. However, the performance and the quality of VoIP is affected by echo. In this paper, a packet based echo canceller (PBEC) is introduced, as a solution to cancel echo in the VoIP network. PBEC can replace the current echo cancellers, which are located in the public switched telephony network (PSTN) central switches. The operating principle of the PBEC is explained and its advantages highlighted. The performance of the PBEC operating using different speech codecs is also studied. Using the PBEC, a maximum ERLE of 37.39 dB has been achieved when used with the pulse code modulation (PCM) based speech codec. From the simulation results, it can be also shown that the performance of the adaptive differential pulse code modulation (ADPCM) clearly matches the performance of the PCM based speech codec.

Subband image coding techniques have received considerable attention as a powerful source coding ones. These techniques provide good compression results, and also can be extended for progressive transmission and multiresolution analysis. In this paper, we propose a hybrid subband image coding scheme using DWT (discrete wavelet transform), DPCM (differential pulse code modulation), and ADPCM (adaptive DPCM). This scheme produces both simple, but significant, image compression and transmission coding.

A description is presented of an effective adaptive encoder, an adaptive differential pulse code modulation (ADPCM) encoder whose main part is a LMS adaptive predictor, and applications in vibration signal compression. Some simulation results are also presented and analyzed. The ADPCM encoder described has the advantage of adapting statistical variation of the input signal and can maintain a relatively high compression ratio in such situations. It is an effective method in vibration signal compression. >

This paper presents a very low power consumption one-chip baseband large-scale integrated circuit (LSIC) for personal communication terminals. It comprises a /spl pi//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.

The authors describe a single-chip, half-duplex multirate transcoder device. The device is pin selectable as a 32-kb/s ADPCM (adaptive differential pulse code modulation) transcoder and a 16-kb/s subband coding transcoder, and it provides a 64-kb/s PCM passthrough option. The transcoder implementation is fully compatible with the 32-kb/s ADPCM standard defined by ANSI and CCITT recommendation G.721 bis. The subband coder (SBC) algorithm provides a high-quality, cost-effective medium bit-rate solution. The ADPCM and SBC implementations on the device produce toll quality speech. >

In this paper, a Hermitian symmetry based orthogonal frequency division multiplexing (HS-OFDM) signal transmitter is proposed. It can be used in a digitized radio-over-fiber (DRoF) system to reduce the system complexity by transmitting a real digital sequence. In this HS-OFDM transmitter, differential pulse code modulation (DPCM) schemes are used instead of the traditional pulse code modulation (PCM) to quantify the bandpass sampled signals, which can improve the signal-to-quantization noise ratio (SQNR) by 4~7 dB. Moreover, a high-order pulse amplitude modulation (PAM) is utilized to increase the spectral efficiency of the DRoF system. Simulation results show that the digital code with PAM- 16 can improve the spectral efficiency by 3 times with the same error vector magnitude (EVM) performance.

To describe and analyse the process of uniform quantization. To describe the principle of pulse code modulation (PCM) as a combination of quantizing, sampling and pulse encoding. To assess analytically the noise performance of a PCM system. To note briefly the merits of non-uniform quantization or companding for encoding certain signals, such as speech. To examine the application of PCM to television and to illustrate the subjective impairment of contouring. To discuss briefly differential pulse code modulation (DPCM) and delta modulation (DM) schemes. To outline the principles of, and note the CCITT recommentations for, PCM-TDM telephony.

A speech-coding algorithm based on the introduction of a vector quantizer into an ADPCM (adaptive differential pulse code modulation) configuration is presented. This vector ADPCM (VADPCM) algorithm is directed toward low-complexity, low-delay (0-5 ms), 16-kb/s applications. An analysis-by-synthesis configuration is used to allow the vector quantizer to operate with the usual scalar linear predictor. Performance/complexity tradeoffs are described. Methods for reducing the implementation complexity to the level of the standard 32-kb/s CCITT algorithm are indicated. Noise feedback, postfiltering, and gain-adaptive vector quantization are used to improve the performance while maintaining low complexity.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>