Richard Louis Zinser

Fading bit error protection for digital cellular multi-pulse speech coder

Protection of a digital multi-pulse speech coder from fading pattern bit errors common in a digital mobile radio channel is accomplished with error detection techniques which are simple to implement and require no error correcting codes. A synthetic regeneration algorithm is employed which uses only the perceptually significant bits in the transmitted frame. Separate parity checksums for line spectrum pair frequency data, pitch lag data and pulse amplitude data are added to each frame of speech coder bits in the transmitter. The bits are then transmitted through a mobile environment susceptible to fading that induces bursty error patterns in the stream. At the receiving station, the parity checksum bits and speech coder bits are used to determine if an error has occurred in a particular section of the bit stream. Detected errors are flagged and supplied to the speech decoder. The speech decoder uses the error flags to modify its output signal so as to minimize perceptual artifacts in the output speech. Separate checksums are developed for subsets of line spectrum pair (LSP) coefficients and related speech data, whereby a single subset may be error-detected and replaced, rather than an entire frame.

Linear predictive codeword excited speech synthesizer

A linear predictive codeword excited speech synthesizer performs a voiced/unvoiced decision to determine the type of excitation to be supplied to a synthesis filter. The synthesizer selects the excitation for voiced speech from a codebook, using an analysis-by-synthesis technique in which the transfer function of a linear predictive coefficient synthesis filter closely resembles the gross spectral shape of the input speech signal. By pitch-periodic repetition of the selected codebook vector, a high quality synthetic speech output is generated.

Some experimental and theoretical results using a new adaptive filter structure for noise cancellation in the presence of crosstalk

The application of adaptive filters in noise cancelling often requires the relative placement of the two transducers at a distance that necessitates a large order filter in order to obtain an adequate output signal-to-noise ratio. A new adaptive filter structure is introduced that permits a closer placement of the transducers and that allows the cancellation of noise in the presence of crosstalk. Algorithms are developed for the new transversal and lattice structures. Simulations show considerable improvement in mean-square error over that obtained with standard noise cancelling algorithms.

An efficient, pitch-aligned high-frequency regeneration technique for RELP vocoders

A technique for reducing the tonal noises due to harmonic misalignment in RELP vocoders is developed. The method, pitch-aligned high-frequency regeneration, is based upon frequency translation by modulation phenomena of the transmitted baseband signal. The method is reported to give the same tonal noise reduction as a full-band pitch prediction system. A reduced complexity version is implemented in a real-time, single TMS320-based vocoder.

CELP coding at 4.0 kb/sec and below: improvements to FS-1016

While code excited linear prediction (CELP) speech coders can produce high levels of output speech quality at rates near 4 kb/s, they may not be suitable for toll quality communications. The authors propose several modifications to improve the quality of the standard CELP algorithm while simultaneously reducing its transmission rate. The modifications include a multimode excitation that increases the convergence rate of the adaptive (pitch) codebook. A low-complexity spectral vector quantization algorithm is also developed that reduces the coding rate and decreases the spectral distortion. SNR improvements of 2 dB and a significant reduction in perceptual artifacts have been observed.<<ETX>>

2.4 kb/sec compressed domain teleconference bridge with universal transcoder

Advanced new technologies, such as cellular-telephone-quality ultra-low-rate speech coders, model domain transcoders, and compressed domain conferencing algorithms provide an opportunity to develop a compressed domain conference bridge system for use in secure, survivable military communications environments. The new conference bridge will allow seamless interoperability with diverse voice terminals and enable full-duplex teleconference operation. Unlike users of half-duplex systems, conferencing participants will be able to talk at the same time and hear the two most relevant simultaneous talkers over a single 2.4 kbit/s connection. This paper describes a system architecture that implements the features mentioned above. Compared to conventional multicast conferencing algorithms, the new system will consume a significantly smaller portion of the satellite resources; for N conference participants, conventional multicast requires N/sup 2/ channels, while the new system will use only 2N channels.

Multiple source MOS evaluation of a flexible low-rate vocoder

This paper describes the design and MOS (mean opinion score) performance of a family of low rate, low complexity speech coding algorithms known as time domain voicing cutoff (TDVC). TDVC is a predictive coding algorithm that employs a single transition frequency dividing voiced and unvoiced excitation. It provides the voicing flexibility of a frequency domain algorithm with lower complexity and rate overhead. A number of algorithm variants were MOS tested using three distinct sets of source material. The results are discussed in terms of performance for each of the three sources, and demonstrate that choice of source material has a great impact on both vocoder scoring and ranking.