Abstract: An improved error control coding scheme is implemented in low bit rate coders in order to improve their performance in the presence of transmission errors typical of the digital cellular channel. The error control coding scheme exploits the nonlinear block codes (NBCs) for purposes of tailoring those codes to a fading channel in order to provide superior error protection to the compressed half rate speech data. For a half rate speech codec assumed to have a frame size of 40 ms, the speech encoder puts out a fixed number of bits per 40 ms. These bits are divided into three distinct classes, referred to as Class 1, Class 2 and Class 3 bits. A subset of the Class 1 bits are further protected by a CRC for error detection purposes. The Class 1 bits and the CRC bits are encoded by a rate 1/2 Nordstrom Robinson code with codeword length of 16. The Class 2 bits are encoded by a punctured version of the Nordstrom Robinson code. It has an effective rate of 8/14 with a codeword length 14.

Abstract: A line spectral frequency (LSF) vector quantizer, having particular application in digital cellular networks (DCN), is provided for code excited linear predictive (CELP) speech encoders. The LSF vector quantizer is efficient in terms of bits employed, robust and effective in terms of performance across speakers and handsets, moderate in terms of complexity, and accommodates effective and simple built-in transmission error detection schemes. The LSF vector quantizer employs a minimum number of bits, is of moderate complexity and incorporates built-in error detection capability in order to combat transmission errors. The LSF vector quantizer classifies unquantized line spectral frequencies into four categories, employing different vector quantization tables for each category. Each quantization table is optimized for particular types of vectors. For each category, three split vector codebooks are used with a simplified error measure to find three candidate split quantized vectors.

Abstract: A digital discontinuous cellular communication system has a transmitter that transmits two frames of data following detection of voice inactivity. A receiver includes a comfort noise generator that uses the two frames of data to output noise to the speaker during period of voice inactivity. The comfort noise generator includes synthesis codebook with samples scaled by actual background noise and excitation codebook with samples filtered and scaled by the background noise that are combined to produce comfort noise having attributes and loudness level of the received background noise prior to interruption of transmission. The scaled signals are weighted to vary the loudness level and spectral attributes.

Abstract: A method for encoding a signal that includes a speech component is described. First and second linear prediction windows of a frame are analyzed to generate sets of filter coefficients. First and second pitch analysis windows of the frame are analyzed to generate pitch estimates. The frame is classified in one of at least two modes, e.g. voiced, unvoiced and noise modes, based, for example, on pitch stationarity, short-term level gradient or zero crossing rate. Then the frame is encoded using the filter coefficients and pitch estimates in a particular manner depending upon the mode determination for the frame, preferably employing CELP based encoding algorithms.

Abstract: A digital discontinuous cellular communication system has a transmitter that transmits two frames of data following detection of voice inactivity. A receiver includes a comfort noise generator that uses the two frames of data to output noise to the speaker during period of voice inactivity. The comfort noise generator includes synthesis codebook with samples scaled by actual background noise and excitation codebook with samples filtered and scaled by the background noise that are combined to produce comfort noise having attributes and loudness level of the received background noise prior to interruption of transmission. The scaled signals are weighted to vary the loudness level and spectral attributes.

Abstract: Code excited linear prediction (CELP) is performed using two voiced and unvoiced sets of windows, each set is used both for linear prediction and pitch determination. The accompanying degradation in voice quality is comparable to the IS54 standard 8.0 Kbps voice coder employed in U.S. digital cellular systems. This is accomplished by using the same parametric model used in traditional CELP coders but determining, quantizing, encoding, and updating these parameters differently. The low bit rate speech decoder is like most CELP decoders except that it operates in two modes depending on the received mode bit. Both pitch prefiltering and global postfiltering are employed for enhancement of the synthesized speech. In addition, built-in error detection and error recovery schemes are used that help mitigate the effects of any uncorrectable transmission errors.

Abstract: In many applications involving the coding and processing of speech signals the relevant applicable codebook is one which may be termed a sparse codebook. That is, the majority of elements in the codebook are zero valued. The searching of such a sparse codebook is accelerated in accord with the present invention by generating auxiliary information defining the sparse nature of the codebok and using this information to assist and speed up searches of the codebook.In a particular method of searching the calculation of the distance between a target vector and a stored codebook vector is enhanced by use of a distortion metric derived from energy terms and correlation terms of the codebook entries. Calculation of these energy and correlation terms is speeded up by exploiting the sparseness of the codebook entries. The non-zero elements (NZE) of the space codebook are each identified and are defined by their offset from a reference point.

Abstract: A sub-band speech coding arrangement divides the speech spectrum into sub-bands and allocates bits to encode the time frame interval samples of each sub-band responsive to the speech energies of the sub-bands. The sub-band samples are quantized according to the sub-band energy bit allocation and the time frame quantized samples and speech energy signals are coded. A signal representative of the residual difference between the each time frame interval speech sample of the sub-band and the corresponding quantized speech sample of the sub-band is generated. The quality of the sub-band coded signal is improved by selecting the sub-bands with the largest residual differences, producing a vector signal from the sequence of residual difference signals of each selected sub-band, and matching the sub-band vector signal to one of a set of stored Gaussian codebook entries to generate a reduced bit code for the selected vector signal.