Abstract: The subtracter performs subtraction processing between a pointer outputted from the pointer register and code words candidate outputted from the code word count storing circuit, and in accordance with whether the result is negative or positive, determines the code words of input data words. Code word candidates stored in the code word count storing circuit are created according to a finite-state transition diagram stored in the state transition storing circuit. An encoder and a decoder are thus made compact and faster.

Abstract: A transmitter has a power calculation unit for calculating the power of a signal. A maximum value detection circuit determines a maximum value of an output of the power calculation unit, and a floating point transformation circuit changes an output of the maximum value detection circuit into a floating point representation. A significant bit extracting circuit extracts significant bits from an output of the floating point transformation circuit and outputs the significant bits as the transmitting signal.

Abstract: A method and apparatus to change compression methods without data loss. A real time system for converting compressing data using one of several compression methods is described. When a change in compression rates is requested, the system changes compression methods while continuing to compress data in real time without data loss.

Abstract: This invention relates to duplex and to half duplex transmission schemes and more particularly to the application of such schemes in TDMA and other multiple access environments. In accordance with a first aspect of the invention, there is provided a method of multiplexing signals transmitted by a number of terminals onto a communications link; the link having a capacity less than the total capacity of the terminals; the method including the step of varying the rate of encoding of said signals transmitted by said terminals whereby the bandwidth o the transmitted signals corresponds to the portions allocated to the terminals. In accordance with another aspect of the invention, there is provided a system for the communication of signals in real time over a channel; said system adapted to vary the bandwidth of at least one signal such that the number of signals multiplexed onto said channel can be dynamically varied.

Abstract: A method and apparatus for encoding input digital data by high-efficiency encoding, a method for transmitting the encoded information and a method and apparatus for reproducing and decoding the transmitted information, are disclosed. For encoding, an input signal is forward orthogonal transformed into spectral signals using a windowing function for forward orthogonal transform having a shape A. For decoding, the spectral signals are inverse orthogonal transformed using a windowing function having a shape B different from the shape A. The degree of concentration of the energy of the spectral distribution may be raised and efficient encoding may be achieved even in case of modified DCT (MDCT) wherein the windowing function needs to satisfy a pre-set constraint.

Abstract: A data communications system includes a standard telecommunications apparatus (1) arranged to transport analogue signals in a form of Adaptive Differential Pulse Code Modulated (ADPCM) digital signals generated from Pulse Code Modulated (PCM) digital signals representative of the analogue signals. A mapping data processor (13) coupled to a source (14) of digital data and to an ADPCM encoder (3) of the standard telecommunications apparatus (1), generates PCM coded symbols representative of the digital data in accordance with a predetermined mapping function. An unmapping data processor (15) connected to an ADPCM to PCM decoder (10) of the apparatus (1) operates to re-generate the digital data from communicated PCM signals.

Abstract: A digital television receiver includes a filter for minimizing NTSC interference with a received signal having a constant symbol rate. An equalizer has two one symbol delays for each tap. A filtered and a non-filtered version of the received signal are applied to a multiplexer at the input of the equalizer, which is operated at twice the symbol rate. The outputs of the multipliers in the equalizer are combined and applied to a demultiplexer. The filtered and non-filtered equalized signals are compared and a selection as to which of the filtered or non-filtered received signals is to be processed is made based upon the comparison.

Abstract: A method for controlling the point of time when a bypass mode operation is begun, based on the format of pulse code modulation (PCM) data. A switching unit, which is coupled between outgoing-end and incoming-end mobile stations, receives signals output from vocoders respectively associated with the outgoing-end and incoming-end mobile stations, thereby checking respective operation modes of the vocoders. Based on the result of the checking, the switching unit controls the vocoders so that communications between the outgoing-end and incoming-end mobile stations can be enabled when both the mobile stations operate in a bypass mode. Accordingly, it is possible to achieve smooth communications without a degradation in speech quality, as compared to conventional communications methods which do not take into consideration operation modes of outgoing-end and incoming-end mobile stations.

Abstract: The invention is a process and apparatus for estimating a bit budget in a circuit for compressing digital data, especially digitized television signals. Signals represented by N-bit coded samples, are estimated based upon the probability of a bit of given order having a given value (0 or 1) out of M binary words representing M samples of the signal. An entropy calculation is performed on the basis of the probabilities corresponding to the N bits with the sum of all the entropies corresponding to the desired estimate.

Abstract: A base station for a radio communication system suitable for use if a radio data communication network is configured using a mobile radio terminal has a base station apparatus for a radio communication system which communicates with a mobile radio terminal over a radio line. The base station apparatus for a radio communication system has a voice communication processing system for processing voice signals and a data communication processing system for processing data signals. The data communication processing system has a low-speed modulating/demodulating unit for conducting a modulating/demodulating process on data signals at a processing speed lower than a processing speed of the voice communication processing system, thereby dividing a way of use of a communication channel and a control channel into data communication purpose and voice communication purpose upon data communication so as to increase an efficiency of utilization of the line.

Abstract: A system and methods for providing data compression of data blocks utilizing time characteristics to increase the transmission time of such data over a communication channel. The underlying principle of data compression using time encoding is that data can be encoded in such a manner that, upon receiving the encoded data at the receiving end of a communication channel, the data content can be retrieved by decoding (decompressing) the compressed data in accordance with the time delays in which data arrives at the receiving end. The present invention employs a method of time encoding for data compression wherein a series of data blocks (e.g., a string of binary data) which are comprised of data units (e.g., bytes or bits) are encoded in such a manner that the following information can be retrieved (i.e., decompressed or decoded) at the receiving end of the communication channel: (1) the content of each data unit can be read (e.g.

Abstract: A data encoding method having a prediction setting process that designates a numerically superior symbol as either a "0" or a "1", and the other one is designated as a numerically inferior symbol. Then a binary bit string composed of "0"s and "1"s is input, and the numerically superior symbol is predicted to continuously repeat for n symbols, where n is set as a prediction bit number. A prediction result output process outputs a prediction correct signal that is either a "0" or a "1" when a prediction is correct for the observed series. The process then moves to an operation to perform encoding of a bit series containing the next n symbols, or otherwise outputs the other signal (i.e., that is not used to represent that the prediction is correct) as a prediction failure signal.

Abstract: A data processing device is programmed to decode and transform a stream of data representing a plurality of subband encoded channels of audio data into one or more channels of PCM encoded data for reproduction by a speaker subsystem. An improved method for decoding and transforming utilizes downmix matrices (1021 and 1022) to form downmixed frequency domain channels in buffers (1031-1034). Only two long DCT transform operations (1041 and 1042) and two short DCT transform operations (1043 and 1044) are needed to transform the downmixed frequency domain channels into a left PCM output (1071) and a right PCM output (1072).

Abstract: Apparatus for simultaneously decompressing and interpolating compressed audio data. The compressed audio data is stored in differential log format, meaning that the difference between each two consecutive data points is taken and the log of the difference calculated to form each compressed data point. To efficiently decompress and interpolate the compressed data, advantage is taken of the fact that addition of logs is equivalent to multiplication of linear values. Thus the log of an interpolation factor is added to each compressed data point prior to taking the inverse log of the sum. An integrator block completes the interpolation and decompression of the data.

Abstract: A pulse code modulation (PCM) unit is connected to a bus in a computer system. Also, connected to the bus is a processor and memory. The PCM unit is connected to transmit and receive telephone lines. The PCM unit reads uncompressed PCM data from the memory over the bus and compresses the PCM data for transmission over the telephone line in a cycle steal operation, wherein the bus is not being used by the processor. The PCM receives compressed PCM data from the receive telephone line, decompresses it, and writes the decompressed PCM data to the memory in a cycle steal operation. In addition, selectivity of either .mu.-Law or A-Law conversion, selectivity of even bit inversion for A-Law conversion, and selectivity between the compressed/decompressed PCM data or raw data is provided on a per channel basis. Selectable linear gain is also provided.

Abstract: A 3-channel encoder and corresponding decoder is disclosed for encoding a 3-channel signal into a transmission signal which is backwards compatible, so that prior art one channel decoders are capable of decoding the transmission signal into a mono signal and 2-channel decoders are capable of decoding the transmission signal into a compatible stereo signal. Further, a 3-channel decoder is disclosed for decoding the transmission signal into a 3-channel signal. Also a 4-channel encoder is disclosed for encoding the 4-channel signal into a transmission signal which is backwards compatible, so that prior art one channel decoders are capable of decoding the transmission signal into a mono signal and 2-channel decoders are capable of decoding the transmission signal into a compatible stereo signal.

Abstract: A method generates a reduced number of values representing a sequence of grouped data values and partitions the reduced number of values by first mapping data values into groups of symbols and then partitioning the resulting stream of symbols. The digits representing the first data value in each group are replaced with symbols from a first alphabet. The most significant digit of the second data value in each group and the sign of that second data value are also represented by a symbol from the first symbol set, while the remaining significant bits of the second data value in each group are represented by symbols from a second symbol set. A stream of symbols which represent a sequence of grouped data values is partitioned into first partition symbol groups and second partition symbol groups. Each first partition symbol group comprises the symbols representing the first data value in each group and also the symbol representing the least significant bit of the second data value which follows.

Abstract: A system for annunciating traffic information includes a plurality of stationary transmitters (42,46) for generating relatively short range, highly directional beam signals (44,48) containing a burst data stream signal (10) to each of a plurality of mobile receivers (50,52,54). The stationary transmitter (42) includes a transmitter (42a) connected to a transmitter antenna (56) for generating the beam signal (44) and to a transmitter control (60) which is connected to a transmitter memory (68) for storing a plurality of compressed data voice messages which are read by the transmitter control and generated as data packet signals (16,20,22) in the burst data stream signal (10).

Abstract: A method and apparatus for encoding an input signal in which the input signal is transformed into frequency components which are encoded by quantization from one encoding unit to another. The information specifying zero encoding units among the encoding units in which encoding is done on the assumption that all frequency components contained in them are deemed to be zero are encoded, is encoded, while the quantization step information of the zero encoding units is outputted without encoding. With the encoding method and apparatus, the number of the encoding bits may be decreased while deterioration in the input signal is prohibited for improving the encoding efficiency.

Abstract: A method and system for efficiently compressing a digital audio signal, wherein the digital audio signal includes a plurality of samples. Each of the samples are separated into a plurality of subbands. A signal to mask ratio (SMR) for each the subbands is predicted utilizing a model of relationships between energy values within each of the subbands and SMR values based on a predetermined psychoacoustic model. A number of are allocated bits in response to the predicted SMR and a preselected bit-rate. Then, each of the subbands are quantized based on the number of bits allocated, wherein the digital audio signal may be efficiently compressed.

Abstract: A new technique for the determination of the masking effect of an audio signal is employed to provide transparent compression of an audio signal at greatly reduced bit rates. The new technique employs the results of recent research into the psycho-physics of noise masking in the human auditory system. This research suggests that noise masking is a function of the uncertainty in loudness as perceived by the brain. Measures of loudness uncertainty are employed to form noise masking thresholds for use in the compression of audio signals. These measures are employed in an illustrative subband, analysis-by-synthesis framework. In accordance with the illustrative embodiment, provisional encodings of the audio signal are performed to determine the encoding which achieves a loudness differential, between the original and coded audio signal, which is less than (but not too far below) the loudness uncertainty.

Abstract: There is provided an encoding and decoding apparatus which can provide a high quality of reproduction of sound. There are provided an encoding section and a decoding section. A block length determining section of the encoding section time-sharingly divides an input signal into blocks having a block length corresponding to the degree of change of the input signal. A frequency band dividing section divides the signal of each block into a plurality of frequency bands. A bit allocating section obtains a masking threshold value-to-noise ratio of each frequency band from the magnitude of the power of each frequency band reflecting a psychoacoustic characteristic, and determines the number of quantization bits to be allocated to each frequency band based on the magnitude of the masking threshold value-to-noise ratio. A quantizing section generates a code string by quantizing the signal while allocating a specified amount of bits to each frequency band based on information from the bit allocating section.

Abstract: A method and an apparatus for encoding a digital audio signal are provided. The method comprises the steps of: mapping the digital audio signal into a plurality of sub-bands and outputting the mapped sub-band signals; allocating a number of bits to each sub-band signal according to human psychoacoustic properties, on the basis of the mapped sub-band signals; compensating each sub-band signal by receiving a transmitted previous quantized error, quantizing the compensated sub-band signals according to the corresponding allocated bit number, and transferring the current quantized error to the next quantization; and forming a transmit bit stream in frame units from the quantized data. Therefore, the quantized error generated during encoding of the audio signal is can be minimized.

Abstract: A coder/decoder for samples of speech or other audio signals with bit rate reduction includes an adaptive prediction filter having filter coefficients which are recursively computed according to the known Levinson-Durbin Recursion. Such computation is based on reflection coefficients derived from correlation coefficients applicable to segments of the audio signal samples, the reflection and prediction coefficients being in N-bit fixed point arithmetic. A k.sup.th reflection coefficient after its first computation is buffered with a minimum accuracy of N+1 bits, and the k-1 previously computed prediction coefficients are recomputed with a limitation of N bits. If overflow occurs, the buffered value is divided, e.g. by 2, after the re-computation is completed and then rounded to N bits. Since rounding occurs only after the new prediction coefficients are derived, the recursion achieves reduced error and is simpler to calculate.

Abstract: An audio data coding method of requantizing digital audio data in accordance with the hearing sense characteristic of the human being, comprises: the steps of calculating a quantization error; calculating a level correction value from the calculated quantization error and a level of an input signal; and carrying out level correction of the digital audio data in accordance with the level correction value. In addition, there is provided an audio data coding apparatus for implementing the above-mentioned audio data coding method.

Abstract: Network bit efficiency for transmitting a limited size data frame over a digital communication system is enhanced by a macro-header encoding mechanism which replaces plural header portions of a data frame sequence with a single macro-header byte. The macro-header may be representative of protocol and signal processing operation fields that would otherwise require a longer overhead sequence as a precursor to data transmission. At the receiver, the macro-code is translated into a predefined sequence of opcodes, parameters and data bytes. Whenever the receiver requires another opcode, parameter or data byte, it initially looks to the macro-code. Otherwise, the necessary byte is obtained from the data frame segment being interpreted.

Abstract: A high efficiency speech coding apparatus which can eliminate repetitions of high efficiency speech coding and decoding processes of a digital speech signal upon inter-office transit switching without employing a special exchange and a transit switching system which can prevent degradation in quality of speech information and increase in encode and decode processing delay upon transit switching are disclosed. The apparatus repetitively inserts a synchronizing signal into an output speech signal to an exchange and supervises an input speech signal from the exchange to detect a synchronizing signal in order to determine whether the destination of connection of the exchange is a subscriber line or another trunk circuit connected by way of another high efficiency coding apparatus.

Abstract: A method for transmitting pieces of synchronized picture data having the same data length of which each include a synchronizing signal, an identifying code, pieces of compressed picture data having different data lengths and an error correcting code in that order is disclosed. To correctly decode each of the compressed picture data, it is required to start a decoding operation from a head address of each of the compressed picture signal. Therefore, in cases where a head address of a piece of compressed picture data exists in a piece of synchronized picture data, an existence information flag for informing the existence of the head address, an address pointer for indicating an address of the head address, and a block number for indicating a position of the compressed picture data having the head address are added just after the identifying code to correctly decode the compressed picture data and other pieces of following compressed picture data in the synchronized picture data.

Abstract: Speech signals coded according to the principle of high-resolution long-term prediction (HLTP) have a high accuracy due to the high-resolution which causes a high complexity. The ordinary LTP method is improved by use of oversampling and determining subsegments Cd lying in a preceding segment which precedes a subsegment to be coded, for which it is the case that the number of samples Dd, expressed in the numbers of samples after oversampling, between the initial time instant of the subsegment to be coded and the initial time instant of a subsegment Cd, fulfills the relation Dd=(D*Ob)/d, in which d=2,3,4 . . . n, where n is a positive integer and where Ob and n are chosen in a manner such that Dd is always an integer. Null sample values produced initially for oversampling are given significance by means of an interpolation technique, at predetermined positions which are situated at a spacing Dd from the original samples in the subsegment to be coded.

Abstract: A system and method for communicating audio data in a packet-based computer network wherein transmission of data packets through the computer network requires variable periods of transmission time.

Abstract: A signal processor for a sub-band coding system includes a selector receiving various data, i.e., sample data, allocation information, synchronization and system information during encoding, changing the various data into appropriate forms for efficient encoding the various data and generating same, a memory temporarily storing various data received from the selector, an operator for selectively scaling and quantizing sample data and dequantizing and descaling encoded data, classifier providing outputting the operated data, allocation information input to the operating means, and system information as encoded data in accordance with the frame format of a general sub-band coding system, or receiving and classifying the encoded data into system information, allocation information and operated data, and a control signal generator for generating control signals for controlling the encoding and decoding of the selector, memory, operator and classifier circuits.

Abstract: A process for recovering, at the reception end, data transmitted over a transmission path for digital data streams. A code-word generator for generating a binary code word producing a Dirac pulse is provided. The binary code word is cyclically shifted to create further binary code words. The binary code word and further binary code words are inverted to create still further binary code words. The data is coded, on the transmission end of the transmission path, with the binary code word, with the further binary code words, and with the still further binary code words. On the reception end of the transmission path, the data is recovered by cross-correlating the data and by utilizing the position of the main maximum of the cross-correlation functions.

Abstract: A low-delay code excited linear prediction (LD-CELP) voice coding systemapplied to a scheme, in which the transmitting side interrupts transmission during the voice-nonactive period and the receiving side generates and outputs a comfort noise during the voice-nonactive period. At the transmitting side, when voice non activity is detected by the voice activity detector, the CN flag indicating the interruption of transmission is sent from the CN flag generator and shortly thereafter a background noise from the LD-CELP encoder is sent, followed by the interruption of transmission. At the receiving side, when the background noise following the CN flag is decoded by the LD-CELP decoder, the internal gain and synthesis filter coefficients are held, which is subsequently used to decode the input from the pseudo-noise generator.

Abstract: A multichannel ADPCM compression and expansion device includes N sets of input and output interfaces; an ADPCM compression and expansion device which expansively converts data of K bits in a form of ADPCM signals into PCM signals of M bits in response to a first control signal and outputs the same and compressively converts data of M bits in a form of PCM signals into ADPCM data of K bits in response to a second control signal; and a control unit which divides one signal transmission slot interval into N sections and successively selects the N sets of input and output interfaces in a predetermined order during respective divided intervals divided into N sections and connects to the ADPCM compression and expansion device, and successively performs, while separating the divided interval into a former half and a latter half, a first control which generates the first control signal and sends the data of K bits to the ADPCM compression and expansion device and then returns the expansively converted M bit data to t

Abstract: Arrangement for generating pulse code modulation values in a telephone set, comprising a microprocessor which includes a working store, in which the microprocessor and the working store are connected by a data bus and an address bus, further including an output circuit for outputting the pulse code modulation values, an output memory connected thereto for storing the pulse code modulation values to be output, the working store and the output memory being incorporated in a single memory circuit. The arrangement is preferably structured in such a way that it comprises an addressing device for generating, in response to the microprocessor's addressing of a random location of the output memory, an address of an output memory location to be read out.

Abstract: A fast discrete cosine transform (DCT) based method of encoding and decoding signals is disclosed. A signal of finite length is decomposed into subsignals of shorter length and an approximate DCT (A-DCT) is computed by keeping only selected subsignals. Computation time decreases relative to the number of subsignals being used to compute the A-DCT, fewer subsignals resulting in a faster computation. For signal compression applications, the computed A-DCT coefficients are quantized and coded. For signal recovery, the encoded A-DCT coefficients are processed through a decoder and the decoded signal is recovered using a direct inverse DCT (I-DCT). In some applications, an approximate inverse DCT (AI-DCT) based on similar decomposition can be used to recover the signal from the encoded data. In image coding applications, the image recovered from the decoded signal exhibits negligible block artifacts compared to conventional JPEG, particularly for low bit rate compression.

Abstract: A variable length encoding/decoding system for use with a layered signal format, and for adaptively providing VLC changes without additional signal overhead, includes a plurality of VLC encoding tables, respective ones of the tables being preprogrammed with VLC codes associated with anticipated signal statistics. The layered signal includes blocks of data subject to common parameters which parameters may be associated with signal statistics. The blocks of data include information indicative of such parameters. Each packet is examined for the common parameter and responsive thereto a default VLC coding/decoding table is assigned. In a further embodiment the VLC table used for each successive codeword is made dependent upon the table utilized for the lastmost codeword and the relative value of the lastmost codeword.