Abstract: In a diversity receiver comprising an AGC section for controlling the gain of a tuner, an equalizer equalizes an outputted signal from a fast Fourier transformer. A reliability calculator calculates the reliability value of each of carriers by an equalized pilot signal obtained from the equalizer. A reliability value corrector corrects the reliability value by outputted information from the AGC section. A carrier selecting/combining section performs one of selecting and weighting combining for the carrier in a branch in accordance with the corrected reliability value. When a Viterbi decoder is provided, the Viterbi decoder weights an output from the carrier selecting/combining section with a new reliability value to perform maximum likelihood decoding. It is possible to prevent a reliability value which does not reflect an actually received power from being calculated as a result of increasing a power by an AGC in spatial diversity for every OFDM or FDM demodulated carrier.

Abstract: An available bandwidth measurement system and method. The available bandwidth of a network path is determined by determining round-trip times of packets utilizing varying bandwidth to a receiving node. An increase in packet round-trip time indicates congestion of a link in the path. In some embodiments, the bandwidth range measured is expanded or decreased to locate a congestion point or to provide greater precision in determining available bandwidth.

Abstract: A system and method of load balancing a network having a plurality of client systems and servers. The load balancing system and method distributes requests from the client systems to different servers on a wide area network by selecting the most optimal server for a specific client request. The client request for a server is received from a client system and the destination point of the client request is resolved in part by a load balancing server selector. The client request is sent to a load balancing server based on the resolved destination point and one server out of a subset of the plurality of servers is selected based on a predetermined criteria, such as network traffic congestion or server load, and a conduit to the client system and the selected server for transmission of the client request is provided.

Abstract: A method and system of combining a plurality of parallel communication channels to emulate a single high-bandwidth communication channel. A continuous stream of packets are grouped as traffic aggregates and assigned to queues associated with the plurality of parallel communication channels. The assignment and reassignment of traffic aggregates to the queues is performed dynamically based on measuring the queue load ratios associated with the lengths of the queues for each of the parallel communication channels. Grouping of existing and future packets as traffic aggregates is based on common attributes shared by the packets such as common source and destination IP addresses.

Abstract: The present invention provides a digital data transmitting/receiving method, a base station unit, and a mobile object terminal unit, which enable collection of information desired by the mobile object terminal among service information distributed by the base station by using a broadcasting channel. The base station transmits data after identification information is added in the header information of the service information provided to the mobile object terminal, and the mobile object terminal has an address showing the identification information. The mobile object terminal analyzes the identification information by indicating the address to identify whether the service information is provided to the terminal.

Abstract: The present invention relates to a speech processing device equipped with both a speech coding/decoding function and a speech recognition function, and is aimed at providing a speech processing device equipped with both a speech coding/decoding function and a speech recognition function by using a small amount of memory.

Abstract: A larger number, L', of delta vectors .DELTA..sub.i (i=0, 1, 2, . . . , L'-1) than the required number L are each multiplied by a matrix of a linear predictive synthesis filter (3), their power (A.DELTA..sub.i).sup.T (A.DELTA..sub.i) is evaluated (42), and the delta vectors are reordered in decreasing order of power (43); then, L delta vectors are selected in decreasing order of power, the largest power first, to construct a tree-structure data code book (41), using which A-b-S vector quantization is performed (48). This provides increased freedom for the space formed by the delta vectors and improves quantization characteristics. Further, variable rate encoding is achieved by taking advantage of the structure of the tree-structure data code book.

Abstract: Speech coding uses a periodic excitation signal source, a non-periodic signal excitation source, a synthesizing unit, a filter, and an error evaluating unit. The synthesizing unit synthesizes a periodic excitation signal and a non-periodic excitation signal output from the above-mentioned sources to generate an excitation sound source signal. The filter regenerates an input speech signal from the excitation sound signal. The error evaluating unit controls the periodic excitation signal source and the non-periodic excitation signal source so that the above-mentioned units output periodic and non-periodic excitation signals which minimize a difference of the regenerated signal from an input speech signal. When the error is minimized, the above-mentioned sound source is supplied to a feedback excitation signal modifying unit.

Abstract: A speech coding and decoding system, where the system is operated under a known code-excited linear prediction (CELP) coding method. The CELP coding is achieved by selecting an optimum pitch prediction residual vector P from an adaptive codebook and the corresponding first gain, and at the same time, selecting an optimum code vector C from a white-noise stochastic codebook and the corresponding second gain. The system of the present invention is implemented by a weighted orthogonalization transforming unit introduced therein. The perceptually weighted code vector AC is not used as in the prior art. Rather, the weighted orthogonalization transformation unit transforms the code vector into a perceptually weighted reproduced code vector AC' being made orthogonal to the optimum perceptually weighted pitch prediction vector AP.

Abstract: To improve the voice quality of a digital mobile communication system such as a car telephone or portable telephone when outdoor background noises are superimposed on voices. To achieve the above object, a voice decoding apparatus comprises noise superimposed part detecting means for discriminating between a noise part containing only noises and a voice part containing voices from signal encoded at a transmission side, voice decoding means for decoding an encoded signal in the voice part into a waveform signal, noise decoding means for decoding an encoded signal in the noise part into a waveform signal, and noise control means for controlling the frequency characteristic of said noise part by controlling said noise decoding means when said noise superimposed part detecting means judges the noise part.

Abstract: A speech signal coding apparatus inputs a pitch period generated by coding a speech signal, and outputs information on a range for said pitch period, together with the pitch period. A speech signal decoding apparatus inputs the pitch period and the above information on the range, and determines whether or not the pitch period is within the range. When the pitch period is determined to be within the range, the speech signal decoding apparatus outputs the above pitch period. When the pitch period is determined not to be within the range, the speech signal decoding apparatus outputs as a pitch period a predetermined value within the range.

Abstract: A speech coding system is provided where input speech is coded by finding via an evaluation computation a code vector giving a minimum error between reproduced signals obtained by linear prediction analysis filter processing, simulating speech path characteristics, on code vectors successively read out from a noise codebook storing a plurality of noise trains as code vectors and an input speech signal and by using a code specifying the code vector. In the speech coding system, the noise codebook includes a delta vector codebook which stores an initial vector and a plurality of delta vectors having difference vectors between adjoining code vectors. In addition, provision is made in the computing unit for the evaluation computation of a cyclic adding unit for cumulatively adding the delta vectors to virtually reproduce the code vectors.

Abstract: A speech coding apparatus includes multipliers and prediction filters which successively process a plurality of signal vectors obtained from an index 2.sup.M and dimension N code book to obtain a reproduced speech signal. Error detectors are provided which find the error between the input speech signal and reproduced speech signal. Evaluators are also provided which calculate the optimum signal vectors giving the smallest errors. The multipliers are connected to a reduced code book, which is constituted of n number of code book blocks of index 2.sup.M/n and dimension N/n (where n is an integer of two or more). There are n number of multipliers, n number of prediction filters, n number of error detectors, and n number of evaluators corresponding to the code book blocks.

Abstract: A gain-shape vector quantization apparatus is provided for encoding and decoding, to transmit and receive compressed speech signals. A selected plurality of vectors are read from a code book based upon an index signal. The vectors are added in an adder and synthesis filtered by a synthesis filter, in either order, to produce an output. This output is subtracted from an input speech signal to produce an error signal. An evaluation unit produces an index to select the plurality of vectors read from the code book memory based on the error signal in order to minimize this error signal. The evaluation unit produces gain adjusting signals which can be used to adjust gains of the vectors read from the code book. In an encoder, signals indicative of the gain adjusting signal and the index signal are transmitted by a transmitter of the encoder to send a quantized speech signal to a receiver of a decoder.

Abstract: A code excited linear prediction (CELP) type speech signal coding system is provided, a code vector obtained by applying linear prediction to a vector of a residual speech signal of white noise is stored in a code book. A pitch prediction vector obtained by applying linear prediction to a residual signal of a preceding frame is given a delay corresponding to a pitch frequency and added to the code vector. Use is made of an impulse vector obtained by applying linear prediction to a residual signal vector of impulses having a predetermined relationship with the vectors of the white noise code book. Variable gains are given to at least the above code vector and impulse vector, a reproduced signal is produced, and this reproduced signal is used for identification of the input speech signal. Thus, a pulse series corresponding to the sound source of voiced speech sounds is created.

Abstract: A speech coding system operated under a known code-excited linear prediction (CELP) coding method. The CELP coding is achieved by selecting an optimum pitch vector P from an adaptive codebook and the corresponding first gain and, at the same time, selecting an optimum code vector from a sparse-stochastic codebook and the corresponding second gain. Special code vectors are loaded in the sparse-stochastic codebook, which code vectors are hexagonal lattice code vectors each consisting of a zero vector with one sample set to +1 and another sample set to -1.

Abstract: Pitch periods for a long term predictor included in a speech codec are searched in two searching stages. In the first searching stage, probable pitch periods are searched skipping a constant number of pitch periods, and in the second searching stage, pitch periods including the pitch period determined in the first searching stage and pitch periods neighboring the pitch period on both sides are searched.

Abstract: A speech coding apparatus coupled to a transmission channel includes m (m is an integer greater than 1) coders, m decoders and m or (m-1) error correcting coders. The apparatus also includes an evaluation unit which evaluates a quality of each of reproduced speech signals from the input speech signal and the reproduced speech signals and which outputs an evaluated quality of each of the reproduced speech signals. The quality of each of the reproduced speech signals is evaluated in a state having no transmission error. A decision unit identifies one of the m coders which provides the reproduced speech signal having a smallest distortion on the basis of the evaluated quality of each of the reproduced speech signals, a current error rate of the transmission channel and error correcting abilities of the error correcting coders, and generates a coder identification number representative of a selected one of the m coders.

Abstract: A CELP type speech coding system is provided with an arithmetic processing unit which transforms a perceptual weighted input speech signal vector AX to a vector .sup.t AAX, a sparse adaptive codebook which stores a plurality of pitch prediction residual vectors P sparsed by a sparse unit, and a multiplying unit which multiplies the successively read out vectors P and the output .sup.t AAX from the arithmetic processing unit. In addition, the CELP type speech coding system includes a filter operation unit which performs a filter operation on the vectors P, and an evaluation unit which finds the optimum vector P based on the output from the filter operation unit, so as to enable reduction of the amount of arithmetic operations.

Abstract: An apparatus for compressing and decompressing speech signals. The encoder or decoder comprises a plurality of codebooks each controlling a plurality of indexed code vectors for a different frequency band. Each of the codebooks is provided with a synthesis filter for reproducing a signal wave shape based on a code vector provided by the corresponding codebook. The encoder or decoder further comprises an adder for computing a sum of signal wave shapes reproduced by the synthesis filters. This arrangement can reduce the memory size of each codebook used for an encoding or a decoding process and an amount of operations of the encoding or decoding process.