The application of algebraic code excited linear prediction (ACELP) coding to wideband speech is presented. An algebraic codebook with a 20 bit address can be used without any storage requirements and, more importantly, with a very efficient search procedure which allows for real-time implementation. The authors describe an efficient procedure for searching such a large codebook deploying a focused search strategy, where less than 0.1% of the codebook is searched with performance very close to that of a full search. High-quality speech at a bit rate of 13 kbps was obtained.>

A general framework is introduced which allows both fast search and freedom in designing codebooks with good statistical properties. Several previously proposed schemes are compared from this viewpoint. A backward filtering formulation is given to show that sparse algebraic codes (SACs) (i.e., with few nonzero components) offer distinct advantages. It is shown that they reduce the optimal-search computation per codeword. They also allow control of the statistical properties of the codebook in the time and frequency domains. This control can be dynamic in the sense that it can be made to evolve as a function of the linear predictive coding model A(z). The algebraic-code excited linear prediction (ACELP) technology which allows full duplex operation on a single TMS320C25 at rates between 4.8 and 16 kb/s and which is based on SAC-driven dynamic codebooks is described.>

The authors present an algebraic code-excited linear prediction (CELP) speech coder where the innovation codebook comes from the first spherical code of the Barnes-Wall lattice in 16 dimensions. Novel fast optimal algorithms for finding the best sequence in this Barnes-Wall shell innovation codebook are described. This algebraic codebook makes it possible to design a CELP coder at 9.6 kb/s with good quality and still implementable on a current digital-signal-processing chip.>

Code-Excited Linear Prediction (CELP) produces high quality synthetic speech at low bit rate. However the basic scheme leads to huge computational loads. The paper describes a related scheme, which allows real time implementation on current DSP chips. The very efficient search procedure in the codebook is achieved by means of a new technique called "backward filtering" and the use of algebraic codes. RSB performances are reported for a variety of conditions.

The following paper discusses a vectorial description of the subband decomposition/ reconstruction process and shows its advantages compared to the usual description. The following points are examined:\bulletnecessary and sufficient conditions for exact reconstruction,\bulletreconstruction error measure,\bulletinteresting properties of exact reconstruction processes, and comparison between transform coding and subband coding.

This paper deals with comparisons between various coding schemes for the digital transmission of high quality sound signals typical of what will be transmitted through a Direct Broadcast Satellite System (DBS). Several PCM coding techniques using instantaneous logarithmic coding laws [1] (A-law and µ-law) or near-instantaneous coding law (block coding scheme) such as the NICAM 3 system [2] as well as the Adaptive Delta Modulation (ADM) [3] are compared in this paper. Several pre-emphasis/de-emphasis laws are also proposed to optimally shape the spectrum of the sound program signal before encoding [1]. An investigation of the properties of these laws is carried out and an evaluation, both objective and subjective, of the impact of their utilisation in a codec is presented.

This paper shows a technique of encoding the LPC residual which allows the achievement of speech coding with residual excitation at a bit rate as low as 2400 bps. The method is inspired by the multipulse coding approach introduced by Atal, associated with an irregular downsampl-ing. The real time implementation of a 4800 bps vocoder on a single TMS 320 DSP is discussed.

For digital modulations applied to the coding of speech signals, a fixed sampling and transmission rate is always chosen. For commercial telephone these rates are respectively, 8 KHz and 64 Kbits/sec, corresponding to a filtered signal bandwidth of 300 to 3300 Hz. A new processing concept (variable sampling and digital quantization) is proposed where a sample is coded with a single binary word. The code word corresponds to an information pair: a variable and adaptive sampling time and a coding angle associated to the signal. The basic principle is conceived around a distribution of the coded samples in amplitude and time (amplitude-time coding) along an adaptive coding curve associated to each coded/decoded sample of the original signal. A variable sampling rate requires a buffer, thus a delay, for transmission at a fixed rate. The transmitted signal so obtained is a transposition of the original speech signal and consequently its characteristics (bandwith, amplitude dynamic range) are modified. Some of the characteristics of the transmitted signal are ultimately used for the digital or even the analog transmission of the signal.

We present in this correspondence a numerical approach to the problem of minimizing the fractional out-of-band power for a class of digital constant envelope modulated signals known as MSKtype signals. The present optimization procedure is much more manageable than a recently suggested analytical one while achieving the same results.

A collection of easily generated signaling waveforms for a binary baseband data system is presented along with the required transmit filter functions. Filtering is based on the criterion of zero intersymbol interference and optimum Peversus receiver input signal-to-noise ratio. The channel considered is a linear one of bandwidth equal to[(1 + \alpha)/2]Hz/baud; α being allowed to vary between zero and one. The results apply, as well, to multilevel data and to bandpass systems, such as digital radio.

For all MSK-type signals proposed until now, the main lobes of their power spectral densities were observed to be always wider than those for conventional PSK. Although it is intuitively believed that this observation can be extended to cover all other MSK-type signals not yet found, no formal proof has been reported yet. The present paper attempts to give such a proof.

The paper present an original numerical algorithm for reducing by a factor of two (32 Kbits/ sec) the bit rate of telephone speech already digitized in standard PCM. At the end of the line (i.e. expensive media) an inverse algorithm resynthesizes a PCM signal which preserves a high telephone quality. This bit-rate compression algorithm is thus essentially "transparent" to PCM transmissions and therefore finds immediate applications for increasing carrier capacity. In particular when used in connection with speech interpolation techniques [1] it enables a fourfold increase in the number of channels per carrier.

It is shown that it is possible to generate a band-limited PSK signal using a ROM and a four-quadrant multiplying D/A Converter.

Methods are presented for the transmission of wideband digital signals above and below the frequency division multiplexing (FDM) baseband spectrum of analog microwave systems. In particular, it is shown that a Data Above Voice (DAV) system employing a 1.544 Mbit/s data channel above FDM channels and a Data Under Voice (DUV) system employing 1.544 Mbit/s channel below 720 FDM channels, are practical hybrid systems. Theoretical models of hybrid transmission are disucssed and the DAV/DUV methods are compared. A conceptual hybrid system design procedure is introduced and field measurement results on an 8 hop 6 GHz system are presented, comparing favorably with theoretical predictions.

A hybrid information carrier is defined as a carrier for partly analog and partly digital source information. Some of the reported schemes transmit 1.5 Mbits/s and 1200 frequency division multiplexing (FDM) channels, others 6.3 Mbits/s and 600 FDM channels, in the shared spectrum of the microwave baseband (BB). Baseband diversity (BBD) is frequently used in this type of system. As hybrid transmission is getting wider popularity, it is important to determine the requirements on hybrid BBD systems. Different block diagrams are considered and performance of realizable hybrid BBD units is discussed. The BBD has different configurations for different data rates. We discuss the reasons for this within the theoretical performance and practical limit of technology. The important functions of signal presence and noise evaluation occur in the BBD. A novel approach to noise evaluation in the BB of microwave systems is feasible with digital techniques. The analog noise and pilot detection are in its threshold for high noise condition and fast pilot sensing time requirement. The proposed scheme evaluates digitally the error probabilityP(e), based on error detections, and works well in high noise conditions.

Control, supervisory, and order wire signal transmission, i.e., service channel (SC) is a requirement for most microwave systems. In digital radio the data spectrum extends to the lowest baseband (BB) frequencies. Frequency translation of the SC above the data spectrum is one possibility. Another method, in frequent use, is the double modulation technique, e.g., the SC amplitude modulates the carrier that is already phase modulated by the data spectrum. A novel approach that results in significant equipment cost savings and good performance is proposed. The low-frequency power density spectrum of the random multilevel channel coded data is removed. The vacated spectrum is available for the SC. For different bit rates, the error probabilityP(e)degradation due to SC is calculated. The analysis is performed for various channel coded messages and SC spectrum utilizations. The high-pass filter that assures low-frequency data spectrum removal is optimized for the SC signal to noise ratio and dataP(e)degradation. The analysis is verified with computer simulations and actual multilevel frequency-shift keying (FSK) 6-GHz microwave hop measurements. The proposed approach is also applicable to analog microwave systems when carrying hybrid information. The hybrid system carries an SC, 6.3-Mb/s data, and 900 frequency division multiplexed voice channels in a shared BB.

Performance evaluations and measurement principles of hybrid information carriers, that is, carriers for partly analog and partly digital source infomation, are established. Simulation requirements, which are independent from a particular hybrid configuration or system capacity, for the analog and digital sources are presented. In order to perform the analysis of multilevel digital streams, a simple formula for the power spectrum density of the randomn-level Gray-coded stream is derived. With normalization of the formula, the same power density spectrum for any number of levels is achieved. The required length of the pseudorandom sequence as a function of the transmission characteristics and hybrid environment is established. The measurement difficulties, when the noise source is not Gaussian, are discussed.