Abstract: A waveform interpolation speech coding apparatus and method for reducing complexity thereof are disclosed. The waveform interpolation speech coding apparatus includes: a waveform interpolation encoding unit for receiving a speech signal, calculating parameters for a waveform interpolation from the received speech signal, and quantizing the calculating parameters; and a realignment parameter calculating unit for restoring a characteristic waveform (CW) using the quantized parameter, calculating a realignment parameter that maximizes a cross-correlation among consecutive CWs for the restored CW.

Abstract: Provided are a flexible bit rate code vector generation method and a wideband vocoder employing the same. This invention implements a flexible bit rate by getting three code vectors which are composed of 24, 16, and 8 pulses, at a time in a search process, through improvement of an algebraic codebook search process in a wideband AMR-WB vocoder. The method includes the steps of: performing a preprocess, wherein the preprocess divides a sub-frame by tracks and decides a pulse position having a maximum value in each track; among a plurality of pulses to be searched, fixing a same number of pulses as the tracks to the position with the maximum value of each track sequentially, and searching optimal positions having a minimum error with a target signal by combining two pulses in two consecutive tracks for the remaining pulses; and creating a code vector with flexible bit rate.

Abstract: The present invention reduces complexity of computation as about 40% comparing to the conventional depth first tree search method. A method for searching an algebraic codebook in algebraic code excited linear prediction (ACELP) vocoding using a depth first tree method, includes the steps of: a) searching branches of predetermined levels to predict a branch in which optimum pulse is located; b) choosing a predetermined number of branches according to the search result of the step a) and removing residual branches; and c) searching the chosen branches and choosing optimum algebraic code.

Abstract: A method for encoding a channel using a parallel convolutional encoder is disclosed. The method includes the steps of a first step for performing an encoding operation using a parallel convolutional encoder at the time when a data is inputted into a frame input data register for implementing a channel encoding operation and generating a code symbol, and a second step for storing the thusly generated code symbol into an interleaver RAM which is alternately selected by the frame unit.

Abstract: An improved pitch searching time reducing method for a CELP vocoder using a Line Spectral Pair (LSP) frequency which is capable of significantly reducing the pitch search time by separating the speech signal using a first formant frequency of the line spectral pair of the digital type personal communication system, which includes the steps of computing a decimation interval of a pitch search interval using an LSP frequency of a first formant computed by a formant filter so as to compute a preparatory pitch of a given speech; determining a preparatory pitch to be used when searching a pitch by detecting a peak and a valley within each decimation interval; and computing a preparatory pitch by adapting a first formant frequency of an LSP computed by a formant filter with a decimation rate and performing a pitch search with respect to the obtained preparatory pitch.

Abstract: The present invention relates to the method to receive a speech signal, to perform a recognition weighting process on it, to synthesize a synthetic speech signal, to calculate an autocorrelation of the synthetic speech signal whose delay is a predetermined value and an autocorrelation whose delay is 0, to divide the square of the former by the latter, to calculate a pitch lag and a pitch filter coefficient by calculating only the part of a positive peak with skipping over the part of a negative peak by using the results from the dividing operation, and to calculate and output the pitch lag and the pitch filter coefficient by repeating the above process Thus, real-time implementation of CELP vocoder can be achieved.

Abstract: A method for processing an input speech signal to be applied to a CELP vocoder has the steps of obtaining preliminary pitch search intervals by a preprocessing autocorrelation expression from a pitch lag of a synthesized speech signal which is synthesized from a residual signal of the input speech signal; computing coefficients of pitch filter with respect to the preliminary pitches; searching a high interval in the autocorrelation; and removing the remaining interval other than the high interval in the pitch lag. Since the present invention proposes a speech processing method which uses only a high interval in autocorrelation of a voice waveform in pitch-searching, and where such a speech processing method is embodied in a CELP vocoder, total computation time of the CELP vocoder can be decreased 37% or more without lowering speech quality. Therefore, a digital signal processor, which is low in price and is slow in speed, can be embodied in a CELP vocoder.