Abstract: A method of implementing a dual-mode audio decoder and filter is provided. The inverse modified discrete cosine transform (IMDCT) method and circuit for a dual-mode audio decoder perform the IMDCT with respect to a signal encoded using either the MPEG or Dolby AC-3 standard by utilizing a shared fast Fourier transform (FFT) circuit thereby simplifying the necessary hardware construction. Also, the number of IMDCT outputs used for windowing is reduced by utilizing the properties of the IMDCT outputs of the MPEG bit stream and thus the size of memory necessary for storing the IMDCT outputs is reduced.

Abstract: The coder/decoder (codec) system of the present invention includes a coder and a decoder. The coder includes a multi-resolution transform processor, such as a modulated lapped transform (MLT) transform processor, a weighting processor, a uniform quantizer, a masking threshold spectrum processor, an entropy encoder, and a communication device, such as a multiplexor (MUX) for multiplexing (combining) signals received from the above components for transmission over a single medium. The decoder comprises inverse components of the encoder, such as an inverse multi-resolution transform processor, an inverse weighting processor, an inverse uniform quantizer, an inverse masking threshold spectrum processor, an inverse entropy encoder, and an inverse MUX. With these components, the present invention is capable of performing resolution switching, spectral weighting, digital encoding, and parametric modeling.

Abstract: The coder/decoder (codec) system of the present invention includes a coder and a decoder. The coder includes a multi-resolution transform processor, such as a modulated lapped transform (MLT) transform processor, a weighting processor, a uniform quantizer, a masking threshold spectrum processor, an entropy encoder, and a communication device, such as a multiplexor (MUX) for multiplexing (combining) signals received from the above components for transmission over a single medium. The decoder comprises inverse components of the encoder, such as an inverse multi-resolution transform processor, an inverse weighting processor, an inverse uniform quantizer, an inverse masking threshold spectrum processor, an inverse entropy encoder, and an inverse MUX. With these components, the present invention is capable of performing resolution switching, spectral weighting, digital encoding, and parametric modeling.

Abstract: A method for effectively suppressing background noise in a speech detection system comprises a filter bank for separating source speech data into discrete frequency sub-bands to generate filtered channel energy, and a noise suppressor for weighting the frequency sub-bands to improve the signal-to-noise ratio of the resultant noise-suppressed channel energy. The noise suppressor preferably includes a subspace module for using a Karhunen-Loeve transformation to create a subspace based on the background noise, a projection module for generating projected channel energy by projecting the filtered channel energy onto the created subspace, and a weighting module for applying calculated weighting values to the projected channel energy to generate the noise-suppressed channel energy.

Abstract: A method of implementing a dual-mode audio decoder and filter is provided. The inverse modified discrete cosine transform (IMDCT) method and circuit for a dual-mode audio decoder perform the IMDCT with respect to a signal encoded using either the MPEG or Dolby AC-3 standard by utilizing a shared fast Fourier transform (FFT) circuit thereby simplifying the necessary hardware construction. Also, the number of IMDCT outputs used for windowing is reduced by utilizing the properties of the IMDCT outputs of the MPEG bit stream and thus the size of memory necessary for storing the IMDCT outputs is reduced.

Abstract: For voice and musical signal, a signal coding system which can obtain good sound quality even at a low bit rate is provided. The signal coding system predicts an input signal in a predicting circuit and performs orthogonal transformation in a first orthogonal transformation circuit of a predicted residual error signal. In a coefficient calculating circuit, a coefficient of a smaller degree is calculated for expressing an envelop of the orthogonal transformation coefficient in the orthogonal transformation circuit. In a quantization circuit, quantization is performed by expressing the orthogonal transformation coefficient with a plurality of pulse trains with determining positions for generating a pulse using the output of the coefficient calculating circuit. The envelop of the orthogonal transformation coefficient is expressed by the coefficient with smaller degree.

Abstract: A method and system for transforming a sampling rate in speech recognition systems, in accordance with the present invention, includes the steps of providing cepstral based data including utterances comprised of segments at a reference frequency, the segments being represented by cepstral vector coefficients, converting the cepstral vector coefficients to energy bands in logarithmic spectra, filtering the energy bands of the logarithmic spectra to remove energy bands having a frequency above a predetermined portion of a target frequency and converting the filtered logarithmic spectra to modified cepstral vector coefficients at the target frequency. Another method and system convert system prototypes for speech recognition systems from a reference frequency to a target frequency.

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 radio communication system includes a voice recognition system (221) for converting (400) a caller's voice message to a textual speech message. The textual speech message is then transmitted to an intended selective call radio (122). To perform these functions, the radio communication system includes a caller interface circuit (218), a transmitter (116), and a processor (222). To perform voice-to-text conversion, the processor is adapted to cause the caller interface circuit to sample a voice signal generated by the caller during a plurality of frame intervals, and to apply a Fourier transform thereto, thereby generating spectral data. The spectral data is subdivided into a plurality of bands. The spectral envelope of the spectral data is then filtered out to generate filtered spectral data. A Fourier transform is applied thereto to generate an autocorrelation function for each band.

Abstract: A speech encoding and decoding system comprises a speech coding apparatus and a speech decoding apparatus. The speech encoding apparatus orthogonally transforms an input speech signal represented in a time domain into a signal represented in a frequency domain in units of predetermined blocks, smoothes the resulting orthogonal transform coefficients by auxiliary information obtained by analyzing the speech signal, vector-quantizes the smoothed orthogonal transform coefficients to generate a quantization index, extracts a vector quantization error of low frequency components of the vector-quantized smoothed orthogonal transform coefficients, scalar-quantizes the vector quantization error to determine low frequency range correction information, and outputs the auxiliary information, quantization index, and low frequency range correction information.

Abstract: The coder/decoder (codec) system of the present invention includes a coder and a decoder. The coder includes a multi-resolution transform processor, such as a modulated lapped transform (MLT) transform processor, a weighting processor, a uniform quantizer, a masking threshold spectrum processor, an entropy encoder, and a communication device, such as a multiplexor (MUX) for multiplexing (combining) signals received from the above components for transmission over a single medium. The decoder comprises inverse components of the encoder, such as an inverse multi-resolution transform processor, an inverse weighting processor, an inverse uniform quantizer, an inverse masking threshold spectrum processor, an inverse entropy encoder, and an inverse MUX. With these components, the present invention is capable of performing resolution switching, spectral weighting, digital encoding, and parametric modeling.

Abstract: The coder/decoder (codec) system of the present invention includes a coder and a decoder. The coder includes a multi-resolution transform processor, such as a modulated lapped transform (MLT) transform processor, a weighting processor, a uniform quantizer, a masking threshold spectrum processor, an entropy encoder, and a communication device, such as a multiplexor (MUX) for multiplexing (combining) signals received from the above components for transmission over a single medium. The decoder comprises inverse components of the encoder, such as an inverse multi-resolution transform processor, an inverse weighting processor, an inverse uniform quantizer, an inverse masking threshold spectrum processor, an inverse entropy encoder, and an inverse MUX. With these components, the present invention is capable of performing resolution switching, spectral weighting, digital encoding, and parametric modeling.

Abstract: A pipelined Fast Hadamard Transform ("FHT") architecture is disclosed that comprises log.sub.2 N identical pipeline stages, which are ideally suited for implementation in application-specific integrated circuits. One 2-port stage of the first illustrative embodiment of the present invention advantageously comprises: a first input for sequentially receiving N incoming correlation signals, I(i), at cycle i, wherein the N incoming correlation signals are based on N Walsh chips; a processor for generating N outgoing correlation signals based on the N incoming correlation signals, I(i); and a first output for sequentially outputting the N outgoing correlation signals, O(i); wherein ##EQU1## k=(i-N/2).

Abstract: A coding process having a band dividing filter and a decoding process having a subband synthesizing filter are arranged so that the operating accuracy is enhanced only in a specific subband, for securing the necessary sound quality with a relatively small amount of the operation. A band dividing control unit is served to derive a signal level of each subband from the output result of a fast band dividing filter and generate a high-accurate operation band specifying command for specifying a subband of a low signal level. A high-accuracy band dividing filter is executed to perform a band dividing operation for the subband of the low signal level specified by the high-accuracy band specifying command.

Abstract: A method of transmitting digitized block coded audio signals includes forming scale factors of the digitized audio signals. The n(k-1) differences are formed from k successively in-time scale factors for each frequency sub-band or for a group of spectral values of the audio signal. The n(k-1) differences are grouped into at least two value classes. New scale factors are selected for each of the n sub-bands or spectral value groups based on a sequence of n(k-1) value classes. Identifying information, including the control information indicating at which locations in the sequence of n(k-1) value classes the selected new scale factors are disposed, is associated with each sequence of n(k-1) value classes. The associated selected new scale factors are assigned to each sequence of the sampled signal values and to the identifying information associated with each sequence of sampled signal values.

Abstract: A method and system for providing an inverse transform for an audio compression decoding algorithm in software precalculates a plurality of identified values; each of which is computationally intensive. The method and system then performs a pre-inverse transform complex multiply utilizing a first portion of the identified values and an array of input coefficients to provide a plurality of intermediate values. Thereafter, an inverse transform complex multiply and a post inverse transform multiply are combined to provide a combined complex multiply operation. The combined complex multiply operation uses a second portion of the identified values and the intermediate values provides the inverse transform. Accordingly, through the use of the present invention, the number of instructions for implementing the inverse transform can be substantially minimized.

Abstract: A transform coder wherein different segments of signal samples to be transmitted to a receiver are encoded using different coding transforms. In order to avoid problems encountered with signal samples forming the transition between successive segments an intermediate transform is used for such transition. This has conventionally required significant transmission overhead, because the selected intermediate transform matrix must be transmitted to the receiver to enable it to decode the transitional signal samples. The invention instead provides weighting factors to indicate the extent to which frequency spectra of the basis functions of the intermediate transform resemble the frequency spectra of the basis functions of the transforms used for the adjoining signal segments. The inverse of the intermediate transform can then be calculated by the receiver from the weighting factors.

Abstract: Samples of an audio signal are converted into grouped frames of transformed transform coefficients, each frame having a plurality of regions (subbands). The power in each region is determined, quantized and encoded. A predetermined number of categorizations are applied to the transform coefficients in the plurality of regions. Each categorization assigns to each one of the regions a selected one of a plurality of different scalar nonuniform quantization step sizes. The step size is selected in accordance with the quantized power in the region and the quantized powers in all the regions. The quantized power encoded transform coefficients and an identification of the selected categorization are variable-length encoded and transmitted to a decoder. During decoding, the power of the transform coefficients in each of the regions is reconstructed from the quantized and encoded power. The selected one the categorizations used by the encoder is determined from the transmitted identification thereof.

Abstract: PONS comprises a transform coder and decoder for discrete time electrical signals, in particular acoustic signals. The PONS coder utilizes an integer coefficient transform coder which is not frequency based, which requires almost exclusively fast integer arithmetic, and which spreads incoming signal energy nearly as evenly as possible among coefficients in the transform domain. The PONS coder also has the property that the magnitudes of transform domain coefficients vary by less than about an order of magnitude, so that the PONS coder dispenses completely with time-varying bit allocation. PONS uses only the quantization step to achieve significant compression. Energy spreading also permits reasonably accurate signal reconstruction when significant numbers of transform coefficients are lost or corrupted.

Abstract: A signal encoding method and apparatus for encoding an acoustic signal, a recording medium having the encoded signals, a method for transmitting the encoded signals, and a signal decoding apparatus for decoding the encoded signals, are disclosed. With the encoding apparatus, an attack portion detection circuit detects an attack portion of the waveform acoustic signal where the waveform elements of the waveform acoustic signal rise sharply. A gain control circuit controls the gain for waveform elements at least upstream of the attack portion using a gain control amount adaptively selected from plural gain control amounts specified by a pre-set power of 2. A forward orthogonal transform circuit transforms the waveform acoustic signal into plural spectral components. A normalization and encoding circuit normalizes, quantizes and encodes the gain control information an plural spectral components.

Abstract: A sound signal processing circuit which independently calculates left and right mask levels of sub-band sound samples. A fast Fourier transform circuit performs a fast Fourier transform on input sound samples, and outputs first power spectrum samples decreased to one-half the input sound samples. A sub-sampling circuit produces a prescribed number of second power spectrum samples by sub-sampling processing of adding power spectrum samples by a prescribed number to make a single spectrum. A mask calculating circuit calculates a mask level of second power spectrum samples by determining a contour expressed in a prescribed unit mask function for every second power spectrum sample as a mask for every power spectrum sample, and adds the masks of the respective power spectrum samples.

Abstract: Injection noise and silence are detected in an input speech signal and an external amplifier is switched on or off based on the detected injection noise or silence. The input speech signal is digitized and a first copy of the digitized signal is preemphasized. After the input speech signal is preemphasized, a predetermined number of Mel-frequency cepstral coefficients (MFCCs) and difference cepstra are calculated for each window of the speech signal. A measure of signal energy and a measure of the rate of change of the signal energy is computed. A second copy of the digitized input speech signal is processed using amplitude summation or by differencing a center-clipped signal. The measures of signal energy, rate of change of the signal energy, the Mel coefficients, difference cepstra, and either the amplitude summation value or the differenced value are combined to form an observation vector.

Abstract: Frequency components are broken into a first signal made up of a plurality of tonal components and a second signal made up of other components. The number of the frequency components making up the tonal components is variable. Tonal signals may be encoded efficiently depending on the manner of distribution of their spectral energy to assure more efficient encoding on the whole. If the signals compression coded in this manner are recorded on a recording medium, the recording capacity may be employed effectively. Also, high-quality acoustic signals may be obtained on decoding signals reproduced from the recording medium.

Abstract: The present invention relates to an adaptive transform acoustic coding circuit capable of ensuring the reliability of decoded audio data. The adaptive transform acoustic coding circuit is employed in a mini disc system, and includes a demultiplexer for inputting and demultiplexing a bit stream of an audio signal applied to the mini disk system; a word reconstruction unit for receiving the output of the demultiplexer, extracting and compressing audio spectrum data; a synthesis unit for extending data compressed at a word reconstruction unit for each of low, middle and high bands of the audio spectrum data and synthesizing the extended data; and an error removing unit for receiving a portion of the output of the demultiplexer and providing error removing data to the word reconstruction unit and the synthesis unit.

Abstract: A speech coding system is shown, which comprises a linear transform unit 50 for executing linear transform on an input signal Si with a predetermined block length Sb and an FFT unit 10, 30 for executing Fast Fourier transform on the input signal Si with two different, i.e., large and small, block lengths, a block length setting unit 20 for calculating a predetermined block length Sb to be set in the linear transform unit 50 according to an FFT signal generated in the FFT unit 10, 30 and setting this block length in the linear transform unit 50, and a coding unit 80 for coding an intermediate signal Sm generated in the linear transform unit 50 to form and output a bit stream So. The FFT unit has a function of selecting a block length used for the Fast Fourier transform among two, i.e., large and small, block lengths according to a continuous portion of the input signal Si.

Abstract: A digital acoustic signal of a time region information is divided into a plurality of sub-band groups, and the number of bits allocated to each data of the sub-band groups is reduced by a reducing number Nr in a quantizing unit according to a maximum value selected among energy values of pieces of data in each sub-band group, so that the digital acoustic signal of a frequency region information is produced. Thereafter, Nr bits are added to each data of the digital acoustic signal of the frequency region information in the inversely quantizing unit, and the data of all sub-bands are combined to produce a reproduced digital acoustic signal of the time region information. Thereafter, a residual signal indicating a difference between the digital-acoustic signal of the time region information and the reproduced digital acoustic signal of the time region information is produced, and the residual signal and the reproduced digital acoustic signal of the frequency region information are multiplexed.

Abstract: In a compact disc having a fixed bit rate, the bit rate is rendered substantially variable for improving the sound quality. Each sample is represented by a fixed length of 16 bits. For a sample with a bit surplus K, its data is substantially represented by upper 14 bits. For the lower two bits of the block K are allocated data of lower two bits of data of a bit deficit block L substantially represented by 18 bits.

Abstract: A signal transforming method and apparatus which improves the quality of a signal, and a recording medium therefore. This method and apparatus alters the frequency components of an acoustic time signal. By altering the frequency components the characteristics of the acoustic time signal is transformed such that its quality is improved. The alteration is such that the difference in magnitude of attributes of frequency components within a substantially critical band are adjusted based on characteristics of auditory sensing.

Abstract: A modulation scheme (600) useful in a voice paging system in which both of two orthogonal modulation components (500 and 510) are used to carry two halves of a single voice message destined for a receiver, or two separate voice messages for a receiver. The single voice message is transmitted in half the time.

Abstract: An encoder apparatus and method for compressing a digital input signal derived from an analog signal to reduce the number of bits required to represent the analog signal with low quantizing noise. In the encoder, a digital input signal derived from the analog signal is divided into frequency ranges. The digital signal in each of the frequency ranges is divided in time into blocks, the time duration of which may be adaptively varied. The blocks are orthogonally transformed into spectral coefficients, which are grouped into critical bands. The total number of bits available for quantizing the spectral coefficients is allocated among the critical bands. In a first embodiment and a second embodiment, fixed bits are allocated among the critical bands according to a selected one of a plurality of predetermined bit allocation patterns and variable bits are allocated among the critical bands according to the energy in the critical bands.