Abstract: An encoding/decoding system employing vector quantization realizes a high quality encoding and decoding with decreased quantizing errors, employing a small sized codebook which faithfully represents each of the inputted waveform vectors. An encoding/decoding system includes an encoding apparatus and a decoding apparatus, each having a codebook for storing information vectors representative of a predetermined number of signal patterns and index that determine the information vectors. The encoding apparatus compares a vector representing an object signal to be quantized with each information vector in the codebook, selects an information vector that is closest to the vector and outputs an index for the information vector. The decoding apparatus obtains an information vector corresponding to the index obtained at the encoding apparatus side by referring to the codebook and decodes the object signal. The codebook utilizes a temporary memory connected thereto.

Abstract: A method for providing an echo power measurement of a transmission line having an echo path delay wherein a frequency domain adaptive filter is converged using samples collected from the transmission line that have been converted from a time domain to a frequency domain. A regression method is used to estimate the magnitude of taps of the frequency bins of the frequency domain adaptive filter that have been determined to be not accurately converged. In addition, a method and apparatus for measuring the echo path delay of a transmission line wherein a predetermined number of first time domain adaptive filters covering predetermined time periods are converged using samples collected from the transmission line that have been subband filtered to calculate a rough estimate of the echo path delay. Then, a second time domain adaptive filter is converged using the non-filtered samples to calculate a final echo path delay.

Abstract: A telecommunications network service overcomes the annoying effects of transmitted noise by a signal processing which filters out the noise using interactive estimations of a linear predictive coding speech model. The speech model filter uses an accurate updated estimate of the current noise power spectral density, based upon incoming signal frame samples which are determined by a voice activity detector to be noise-only frames. A novel method of calculating the incoming signal using the linear predictive coding model provides for making intraframe iterations of the present frame based upon a selected number of recent past frames and up to two future frames. The processing is effective notwithstanding that the noise signal is not ascertainable from its source.

Abstract: A sample speech is analyzed by a speech analyzing unit to obtain sample characteristic parameters, and a coding distortion is calculated from the sample characteristic parameters in each of a plurality of coding modules. The sample characteristic parameters and the coding distortions are statistically processed by a statistical processing unit to obtain a coding module selecting rule. Thereafter, when a speech is analyzed by the speech analyzing unit to obtain characteristic parameters, an appropriate coding module is selected by a coding module selecting unit from the coding modules according to the coding module selecting rule on condition that a coding distortion for the characteristic parameters is minimized in the appropriate coding module. Thereafter, the characteristic parameters of the speech are coded in the appropriate coding module, and a coded speech is obtained. When the coded speech is decoded, a reproduced speech is obtained.

Abstract: A method is disclosed for generating frame voicing decisions for an incoming speech signal having periods of active voice and non-active voice for a speech encoder in a speech communication system. The method first extracts a predetermined set of parameters from the incoming speech signal for each frame and then makes a frame voicing decision of the incoming speech signal for each frame according to a set of difference measures extracted from the predetermined set of parameters. The predetermined set of extracted parameters comprises a description of the spectrum of the incoming speech signal based on line spectral frequencies ("LSF"). Additional parameters may include full band energy, low band energy and zero crossing rate.

Abstract: A noise reducing method and apparatus for reducing quantization error or noise generated on quantizing input signals, such as digital audio signals, wherein a dither signal concentrated in signal energy in the low frequency range is summed to the input signal and the resulting sum signal is sent to a quantizer. The quantizer error generated in the quantizer is fed back via a noise filter to the input side of the quantizer. The resulting signal has a smoothed noise floor and has noise components suppressed in a frequency range to which the human hearing mechanism exhibits high sensitivity, thus realizing the sound quality comparable to that of the input signal which prevailed prior to noise shaping.

Abstract: A method and apparatus for reducing noise in an input speech signal, in which the apparatus includes a noise reducing unit that has a variable noise reducing amount based on a control signal that is responsive to the detection of a consonant portion in the input speech signal, such that filter characteristics controlled by the consonant detection are based on a first value found on the basis of a ratio of the input speech signal spectrum and an estimated noise spectrum and a second value found on the basis of a maximum value of the ratio of the signal level of the input signal spectrum to the estimated noise spectrum.

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 speech signal transmitting receiving apparatus, such as a portable telephone set, includes a speech signal transmitting encoding circuit, a noise domain detection unit, a noise level detection unit and a controller. The speech signal transmitting encoding circuit compresses input speech signals by digital signal processing at a high efficiency. The noise domain detection unit detects the noise domain using an analytic pattern produced by the speech signal transmitting encoding circuit. The noise level detection unit detects the noise level of the noise domain detected by the noise domain detection unit. The controller controls the received sound volume responsive to the noise level detected by the noise level detection unit.