Packet Based Echo Cancellation for Voice Over Internet Protocol

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.

Modems designed for the public switched telephone network (PSTN) have conventionally been based on modeling assumptions which view the PSTN connection as an essentially analog medium. However, as the PSTN evolves toward all digital transport and switching, and particularly as major traffic sources, such as Internet service providers, increasingly have direct digital connections to the PSTN, it is appropriate to revisit the model assumptions. Recently, several modem and chipset manufacturers have announced "56 K" modems based on an emerging system paradigm in which one user (a residential Internet subscriber) has an analog connection to the PSTN, and the other (an Internet service provider) has a digital one. ITU-T is expected to finalize details of a corresponding recommendation in 1998. With this configuration, modem designs based on signaling with the /spl mu/-law alphabet become feasible, and the conventional Shannon limit disappears as the quantization noise is avoided. Thus, the conventional Shannon limit of about 36 kb/s can be beaten, and it is possible to approach the digital transmission rate of 64 kb/s. Modems employing this general approach have become known as /spl mu/-law or pulse-code modulation (PCM) modems. In this paper we present a signaling technique and the sampling theory based on this technique, display the structure and operating principles of a PCM modem equalizer, and show how this equalizer problem can be cast in the language of multiinput-multioutput (MIMO) system theory.

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.

This chapter is concerned with different types of modulations: Pulse Amplitude Modulation, Pulse Width Modulation, Pulse Position Modulation, Pulse Code Modulation, Line Codes, Delta Modulation, \(\Sigma -\Delta \) Modulation, Differential Pulse Code Modulation and Adaptive Differential Pulse Code Modulation. In this chapter, Time Division Multiplexing is also explained.

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.

One of the most common ways, where anyone can communicate with each other very easily, and instantly, is, of course, the voice exchange. Public Switched Telephone Networks (PSTN), and Voice over Internet Protocol (VoIP) are common ways of transferring voice communications by using telephone network. These two ways have their own strengths and weaknesses. The two major positive points of the PSTN are reliability, and stability. One of the drawbacks of the PSTN is that it has complex connectivity. The VoIP strengths are cheaper in costs compared to PSTN, less usage of cable, user-friendly, simple connectivity, and variety of services. On the other hand, the biggest challenge of VoIP is frequent disconnects of voice communication. In this paper, a survey is being performed on a selected group of people to understand the quality differences between PSTN, and VoIP. This paper shows that the Quality of Service (QoS) of today's VoIP is not good enough but users still continue to use it. However, this survey is also trying to find ways on how to improve existing QoS of VoIP. Plus, it points out what type of users prefers to utilize by using PSTN or VoIP.

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, 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.

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>

The ubiquitous real-time monitoring and recording of surface electromyography (sEMG) signal is essential to several rehabilitation applications, such as muscle recovery analysis. We present an inexpensive wireless sEMG sensor using a commercial off-the-shelf wireless microcontroller unit (MCU) incorporating a simplified adaptive differential pulse code modulation (ADPCM) routine for real-time data compression. In single-channel configuration, the presented approach reduces power consumption of a transmission subsystem by up to 69%, leading to longer operation life expectancy. Due to the excessive amount of data as well as the limited processing power of embedded MCUs, multi-channel configurations would normally not be feasible. However, the proposed compression method makes a multi-channel EMG sensor possible. The distortion induced by this approach on EMG signal is on the order of 1%. Test results from in vivo trials with humans are presented.

In recent years, internet protocol (IP) has become a good choice over public switched telephone network (PSTN). VoIP implementation uses hard IP phones, soft IP phones and uses Softswitch for call signalling. The Erlang B model is used to determine number of trunks in circuit switched network and found traffic intensity and grade of service (GoS). This paper utilises an extension of Erlang B model for traffic engineering VoIP, i.e., extended Erlang B model. The main purpose for extended Erlang B is that it has better efficiency to handle the percentage of blocked calls by choosing a threshold value (∝). We propose a new measurement scheme based on extended Erlang B model using FreeSWITCH and simulated and analysed VoIP traffic. We compare our version with the original definition of EB model and present further results from simulations. The proposed scheme is also analysed for other QoS parameters, i.e., jitter, end to end delay and MOS.

In personal communication systems, better voice quality and lower terminal price are essential requirements, and adaptive differential pulse-code modulation (ADPCM) codecs are one of the voice codec candidates that have good voice quality and reasonable cost. However, in slow fading environments, burst errors induce click noise in ADPCM voice signal transmission. This paper proposes a novel error-concealment scheme to improve the ADPCM voice signal transmission quality of time division multiple access time division duplex (TDMA-TDD) personal communication systems in slow fading environments. The proposed scheme employs an improved error-detection technique with PCM differential detection and overflow detection coupled with a cyclic redundancy check (CRC) code. To compensate for residual click noise, an expanded CRC error-detection signal is used. Erroneous PCM signals are adaptively suppressed according to the click-noise detection information. Hardware simulation results show that these techniques significantly improve ADPCM voice signal transmission quality by 0.5 points and 1.0 compared to that of the conventional ADPCM muting scheme and no processing, respectively, with the mean opinion scoring (MOS) test in the slow fading environments typical of personal communication systems.

Chapter 11 - Waveform Quantization and Compression