Digital Voice Transmission Research Articles

The shortwave digital broadband radio communication device

For the narrowband analog signals transmission via the radio channels in areas with difficult terrain the short-wave range is widely used. The majority of radio stations use frequency modulation of the transmitter, without any message encryption. This gives the opportunity to detect and intercept messages that is transmitted. The article proposes the use of a digital voice transmission method using a broadband radio signal. The calculated signal/noise ratio, which is created by a broadband transmitter on the input of the radio signals monitoring receiver, in fifty times less than for existing narrowband means of the short-wave range.

Calling All Phones: From a Single-Chip Mobile Communicator to the Smart Phone of the Future

Generally seen as an innovative educational toy, the Speak & Spell played a critical role in the introduction of cost effective and power-efficient voice digitization. Speak & Spell was one of the earliest handheld, battery operated consumer devices having a keyboard and display and produced audio from digitally synthesized sound stored in replaceable cartridge ROM and played through a special digital signal processing clip. Such a technology stood as a precursor to digital voice transmission and ultimately spurred the exponential growth of digital cellular by influencing subsequent technological innovations and trends.

An Investigation of the Perceptual Value of Voice Frames

It is well known, the voice segments and coincident data packets are not equally valued and significant for decoding and comprehension of speech signal. Some lost segments may only slightly worsen audible quality, while others cause strong distortion of the speech signals. Despite this, the feature of different importance of different voice segments in current generation of digital voice transmission systems is not fully used. There is a fundamental problem with discrimination of different importance and value of voice frames. In this paper the concept “of value of voice frame” is introduced, the metric and means for evaluation and measurement of voice frame value are proposed and also results of the measurements of voice frames value are presented.

Validation of a HF spread spectrum multi‐carrier technology through real‐link measurements

AbstractThe HF band data communication link has been traditionally desired by many of the large range transmission systems although it is associated to unfavourable performances as low transmission rate, large delay and low confidence in terms of link establishment and maintenance. Although transmission rates may be high enough to transmit digital voice, delay, usually over several seconds, has been the main handicap to let the systems provide interactive digital voice links. Indeed, there is no unclassified equipment with this capability. The main achievement of this proposal is that we are able to guarantee digital voice transmission with low latency, around 135 ms (modem+codec), providing a full interactive digital voice link. Performances of two new 2460 bps HF modems are presented versus the 39‐tone 2400 bps MIL‐STD‐188‐110A modem, working over an ITU‐R moderate channel. Furthermore, these results are corroborated by real tests carried out in a 1800 Km link in the 18 MHz band. Copyright © 2006 AEIT.

Trellis coded modulation for mobile-satellite communication using space diversity

Most of the past studies concerning the decoding performance of trellis coded modulation (TCM) in a fading channel have been based on time diversity using interleaving/deinterleaving. One problem with time diversity is that a large decoding delay is produced due to the interleaving/deinterleaving which depends on the fading interval. This paper considers TCM signals using space diversity which can drastically improve the decoding delay and analyzes the decoding characteristics in Rice fading channels. An analysis is also made of the effect of the Euclidean distance (ED) and the Hamming distance (HD) of the trellis code on the average bit error rate (BER), and it is shown that a code design considering those two distance structures is needed in a channel with severe fading. This paper considers the major application of the proposed method to be digital voice transmission in mobile communication via satellite. Consequently, a simulation is made for the second-order space diversity decoding performance for the trellis coded 8-PSK signal. The optimal trellis coding at the transmitter is discussed for BER = 10−4.

Second generation wireless information networks

To enhance the capacity and capabilities of wireless information networks, five standardized second-generation networks will be deployed in Europe, North America, and Japan. All are based on digital voice transmission and improved network control relative to the first generation. The author describes the motivation behind the standards and presents details of transmission technologies and logical control channels. Specifically, a system operation is discussed in the context of the first two layers of the open systems interconnection model. The physical layer, the multiple-access technique, the modulation format, and the method of speech coding are discussed. Regarding the data link layer, the information formats of the logical channels that convey control information to and from wireless terminals are presented.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Subjective quality quantification of digital voice communication systems

The expanding digitisation of the telephone networks has brought about an increasing interest in methods addressing the performance characterisation of new digital voice transmission systems. In the paper a review of subjective methodologies for the assessment of telephone quality digital voice communications systems is provided. Technical aspects concerning the network applications and other characteristics relevant to the type of system under evaluation are considered first, since these factors influence the selection of a suitable assessment methodology. Next, listener opinion and conversational opinion tests are described, with particular emphasis on their respective merits, characteristics and differences. Finally, articulation and diagnostic tests, while not generally employed for the assessment of telephone quality, are briefly examined, since the results obtained from such tests are often correlated with the perceptual performance of telephone quality communications links.

Digital voice transmission having improved echo suppression

A digital voice communication apparatus having improved echo suppression includes a transmission gain detection circuit for detecting a near talker signal and a receiver gain detection circuit for detecting a far talker signal. An echo detecting circuit detects which of the near talker and far talker signals has a lesser value and suppresses the gain of the one detected to have the lesser value. A far talker suppressing circuit detects whether the most recent peak gain in this near talker signal is greater than a threshold and is greater than the gain of the far talker signal. If this condition exists the far talker signal is suppressed. A near talker signal suppressing circuit detects whether the most recent gain of the far talker signal is greater than a threshold and is greater than the gain of the near talker signal. If so, the gain of the near talker signal is suppressed.

Error correction/masking for digital voice transmission over the land mobile satellite system

The development of a 2400-b/s speech digitizer which provides an acceptable level of intelligibility and quality over land mobile satellite channels is described. Performance tests over simulated channels in the UHF band (800 MHz) are presented. The voice digitizer is a linear prediction (LPC) vocoder which uses a channel error correction and concealment procedure tailored to error statistics for a minimum-shift keyed (MSK) downlink to a moving vehicle. The error-handling technique is based on perceptual criteria and utilizes the parametric nature of LPC representation of speech. A single-error-correcting, single-burst-detecting (28, 20) fire code is shown to be the best choice for the application. The intelligibility of the vocoder is measured and compared to the standard LPC-10 algorithm. The major remaining sources of speech quality degradation due to channel errors are determined and ranked.< <ETX xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">&gt;</ETX>

Techniques for medium-speed data transmission over HF channels

The paper is a review of techniques for the transmission of medium-speed data over ionospheric channels. Typical applications are digital voice and high integrity data transmission using normal voice bandwidth circuits. The characteristics of the fading multipath propagation channel are described, as well as the relevant radio equipment limitations. An historical overview of modem designs is included from the early multitone designs to the present day. The two basic design approaches of multiple low-speed channels and a single high symbol rate channel with adaptive processing are described. The impact of new digital signal processing technology on modem implementation is noted. The performance of the different designs is compared, both on theoretical grounds and from the results of practical measurements. It is concluded that the single tone approach generally offers superior performance in terms of lower bit error rates, but the choice is highly dependent on the application.

Evaluation of 16 kbit/s digital voice transmission for mobile radio

In this paper 16 kbit/s digital voice transmission with conventional channel spacing of 25 kHz, employing a 16 kbit/s adaptive delta modulation (ADM) coder-decoder (CODEC) is evaluated. The main characteristics of narrow-band digital FM modulation schemes, such as tamed FM, Gaussian filtered minimum shift keying (GMSK), four-level FM and phase locked loop-quaternary phase shift keying (PLL-QPSK), are compared by laboratory tests. Digitized voice quality in a digital channel incorporating a 16 kbit/s ADM CODEC and GMSK coherent detection was compared with voice quality of a conventional analog FM channel. Bit error ratio (BER) performance is shown to depend primarily on demodulation schemes. Digital voice quality is inferior to that of analog voice with an opinion score difference of about 0.5 in fading environments. This kind of digital voice transmission will be applicable for those systems that require high security at an expense of speech quality.

Peripheral board controller for digital exchange systems

The controller implements a redundant PCM highway port and a 4-Mb serial communication link. Typical features are variable time slot assignment for up to 16 subscribers, processing of status signaling and feature control data, and high level support of the HDLC (high-level data link control) communications protocol. The circuit represents an innovative step in the development of modern subscriber boards in digital switching systems, particularly in the area of fully digital voice and data transmission capability. The circuit is realized in a 3.5-/spl mu/m NMOS process and contains more than 22000 transistors on a 7.24/spl times/5.24 mm die area.

A Distributed Experimental Communications System

The packet experimental communications system (packet XCS) is a new experimental voice and data switch. It uses a local-area network (LAN) for digital voice transmission, with local intelligence for switching. The packet XCS also has highly distributed control. The individual sites cooperate to provide user services as well as internal data management. We have learned that several local networks, including CSMA/CD networks, can be made to work well for voice transmission and that highly distributed control is practical in such a system. A system has been constructed which is used as a testbed for distributed voice and data communications experiments. This system is purely for experimentation and does not indicate a direction for future Bell System product offerings.

Satellite repeater system for advanced aircraft/ground communications

This paper considers the application of satellite relay techniques to provide communication paths for aircraft-ground communication over the ocean. It considers the feasibility of a communication system design for air traffic control (ATC) utilizing the reflex repeater as the basic satellite-borne relay device. Implementation of satellite relay techniques would permit continuous line-of-sight operation for reliable ATC digital data links and voice transmission. The paper describes the basic use of the relay system, the repeater electronics, and the communication link parameters required to accommodate the satellite-aircraft link geometries. Examples of typical aircraft-ground link calculations are presented, and the basic supersonic aircraft communication requirements and satellite contact profiles are discussed.

Reduction in quantizing levels for digital voice transmission

Reduction in quantizing levels for digital voice transmission