Abstract: A pitch lag coding device and method using interframe correlation inherent in pitch lag values to reduce coding bit requirements. A pitch lag value is extracted for a given speech frame, and then refined for each subframe. For every speech frame having N samples of speech, LPC analysis and vector quantization are performed for the whole coding frame. The LPC residual obtained for each frame is then processed such that pitch lag values for all subframes within the coding frame are analyzed concurrently. The remaining coding parameters, i.e., the codebook search, gain parameters, and excitation signal, are then analyzed sequentially according to their respective subframes.

Abstract: A signal compression system includes a coder and a decoder. The coder includes an extract unit for extracting an input feature vector from an input signal, a coder memory unit for storing a predesigned vector quantization (VQ) table for the coder such that the coder memory unit uses a set of primary indices to address entries within the pre-designed VQ table, a coder mapping unit for mapping indices from a set of secondary indices to the first set of indices, and a search unit for searching for one index out of the set of secondary indices, wherein the index from the set of secondary indices corresponds to an entry in the coder memory unit, and the entry best represents the input feature vector according to some predetermined criteria.

Abstract: A pitch estimation device and method utilizing a multi-resolution approach to estimate a pitch lag value of input speech. The system includes determining the LPC residual of the speech and sampling the LPC residual. A discrete Fourier transform is applied and the result is squared. A lowpass filtering step is carried out and a DFT on the squared amplitude is then performed to transform the LPC residual samples into another domain. An initial pitch lag can then be found with lower resolution. After getting the low-resolution pitch lag estimate, a refinement algorithm is applied to get a higher-resolution pitch lag. The refinement algorithm is based on minimizing the prediction error in the time domain. The refined pitch lag then can be used directly in the speech coding.

Abstract: A method for efficient coding of non-active voice periods is disclosed for a speech communication system with (a) a speech encoder, (b) a communication channel and (c) a speech decoder. The method intermittently sends some information about the background noise when necessary in order to give a better quality of overall speech when non-active voice frames are detected. The coding efficiency of the non-active voice frames can achieved by coding the energy of the frame and its spectrum with as few as 15 bits. These bits are not automatically transmitted whenever there is a non-active voice detection. Rather, the bits are transmitted only when an appreciable change has been detected with respect to the last time a non-active voice frame was sent. To appreciate the benefits of the present invention, a good overall quality can be achieved at rate as low as 4 kb/s on the average during normal speech conversation.

Abstract: A conventional CELP speech codec synthesizes a pitch interval in a sound by synthesizing a scaled innovation signal--typically, a random signal--and adding it to a scaled pitch signal derived from the synthesized speech of the previous pitch interval. This invention continues this practice when it is advantageous, but, at the onset of the sound or whenever else needed, replaces the scaled innovation signal with a scaled spike signal. This is done since a spike is sometimes more useful than an innovation signal is, innovation signals being by definition crafted to instead represent differences between adjacent pitch intervals within a sound rather than at the onset of a sound.

Abstract: The process for detection and masking of bad frames in a coded speech signal resulting from channel transmission errors has been improved. The coded speech signal has a first group of bits comprising the most perceptually significant bits of the coded speech signal, a second group of bits comprising the second most perceptually significant bits and a third group of bits comprising the least perceptually significant bits. The coded speech signal is de-interleaved to obtain a first series of bits comprising the first and second group and a second series of bits comprising the third group of bits. The first series of bits are convolutionally decoded for recovering the first and second group of bits, with the first group of bits containing error protected bits. A CRC check is performed by sending the recovered bits of the first group to cyclic redundancy decoding means. The first group of bits is forwarded to speech decoder means if the CRC check is successful.