David A. Heide

Speech enhancement for bandlimited speech

Throughout the history of telecommunication, speech has rarely been transmitted with its full analog bandwidth (0 to 8 kHz or more) due to limitations in channel bandwidth. This impaired legacy continues with tactical voice communication. The passband of a voice terminal is typically 0 to 4 kHz. Hence, high-frequency speech components (4 to 8 kHz) are removed prior to transmission. As a result, speech intelligibility suffers, particularly for low-data-rate vocoders. In this paper, we describe our speech-processing technique, which permits some of the upperband speech components to be translated into the passband of the vocoder. According to our test results, speech intelligibility is improved by as much as three to four points even for the Department of Defense-standard mixed excitation linear predictor (MELP) 2.4 kb/s vocoder. Note that speech intelligibility is improved without expanding the transmission bandwidth or compromising interoperability with others.

Acoustic noise reduction for speech communication: (second-order gradient microphone)

Ambient acoustic noise is a significant factor for degrading speech quality of the telephone, particularly, of a digital telephone operating at low data rates. In extremely noisy environments (e.g., flight deck, high-performance aircraft, tanks, helicopters, stock exchange floor, to mention a few), speech communication would be difficult without using some form of noise reduction at the front-end of the telephone. Over the years, therefore, various noise reduction techniques have been developed to improve voice communication in noisy environments. Previous noise reduction methods focused primarily on digital preprocessing of the corrupted speech signal after it has been induced by the microphone. In this paper, we present an alternative that reduces noise while it is being mixed with speech to become corrupted speech. The noise reduction technique is based on the microphone array that is very insensitive to far-field sounds (i.e., noise), while it provides a flat frequency response for near-field sounds (i.e., speech). The microphone array is small enough to be installed within a handset or headset. The prototype we fabricated has shown excellent performance. In an extremely noisy environment where one cannot hear an adjacent person talking, the microphone array discussed in this paper provided highly intelligible speech.

Variable Data Rate Vocoder Improvements for Secure Interoperable DoD Voice Communication

One intermediate goal towards achieving universal secure voice interoperability in the Department of Defense (DoD) has been the development of a universal voice encoder (vocoder). This vocoder would seamlessly encode speech at a wide range of interoperable variable and fixed data rates to suit a wide range of DoD communication equipment. This paper describes the most important features of such a vocoder and recent advancements in achieving this goal. Specifically this paper will describe three main areas: 1) Summary of the variable data rate (VDR) algorithm and the most recent improvements to the vocoder. 2) Methods for transcoding between fixed rate and variable data rate vocoding modes. 3) Summary of heavily bit error protected, fixed-rate modes.

Comparison of TSVCIS voice at 8000 and 12000 bps VERSUS CVSD at 16000 bps

In this paper a novel technique for implementing heavily bit error protected 8 and 12 kilobits per second (kbps) voice coder (vocoder) based on a 2.4 kbps vocoder specified in Tactical Secure Voice Cryptographic Interoperability Specification (TSVCIS) is compared to the current 16 kbps Continuously Variable Slope Delta Modulation (CVSD) used in several legacy military radios and the Single Channel Ground and Airborne Radio System (SINCGARS). TSVCIS voice significantly outscored CVSD in terms of quality, using the Mean Opinion Score (MOS) and Diagnostic Acceptability Measure (DAM), and intelligibility, from the Diagnostic Rhyme Test (DRT). These tests show that TSVCIS voice delivers significantly higher quality and intelligibility than CVSD when transmitted over noisy communication channels.

A novel software defined radio relay method for power conservation

To meet beyond line of sight (BLOS) communications requirements, a novel software defined radio relay method for power conservation is proposed. This method is able to achieve approximately 7.5 watt per relay node power savings for SNR challenged links (probability = 1/3), adjacent interference links (probability = 1/3), and 33% clear links (probability = 1/3). This is accomplished by dynamically adjusting the relay methodology between a low power amplify-and-forward (AF) relay method, a compress-and-forward (CF) relay method, and a power intensive decode-and-forward (DF) relay method. To demonstrate the novel architecture the following models were developed in MATLAB/Simulink and tested on an Avnet Zynq-7000 software defined radio (SDR) with Military RT 1439A radios: an audio FM transmitter model, an audio FM receiver model, an AF relay model, a CF relay model, a DF relay model, and the novel SDR relay model.