Linear predictive coding

About: Linear predictive coding is a research topic. Over the lifetime, 6565 publications have been published within this topic receiving 142991 citations. The topic is also known as: Linear predictive coding, LPC.

Codebook structure and search for speech coding

Abstract: A speech compression system with a special fixed codebook structure and a new search routine is proposed for speech coding. The system is capable of encoding a speech signal into a bitstream for subsequent decoding to generate synthesized speech. The codebook structure uses a plurality of subcodebooks. Each subcodebook is designed to fit a specific group of speech signals. A criterion value is calculated for each subcodebook to minimize an error signal in a minimization loop as part of the coding system. An external signal sets a maximum bitstream rate for delivering encoded speech into a communications system. The speech compression system comprises a full-rate codec, a half-rate codec, a quarter-rate codec and an eighth-rate codec. Each codec is selectively activated to encode and decode the speech signals at different bit rates to enhance overall quality of the synthesized speech at a limited average bit rate.

Method and apparatus for processing speech signals

Abstract: Processing speech signals applicable to a variety of speech processing including narrowband, mediumband and wideband coding. The speech signal is modified by a normalization process using the envelope of the speech signal such that the modified signal will have more desirable characteristics as seen by the intended processing algorithm. The modification is achieved by a point-by-point division (normalization) of the signal by an amplitudes function, which is obtained from lowpass filtering the magnitude of the signal. Several examples of normalized signal are presented. Application to pitch detection and speech coding are described herein.

Comparison of speech parameterization techniques for the classification of speech disfluencies

Abstract: Stuttering assessment through the manual classication of speech disuencies is subjective, inconsistent, of the 2 parameters (LPC order and frame length) in the LPC- and PLP-based methods on the classication results is also investigated. The experimental results reveal that the proposed method can be used to help speech language pathologists in classifying speech disuencies.

Variable-rate CELP based on subband flatness

Abstract: Code-excited linear prediction (CELP) is the predominant methodology for communications quality speech coding below 8 kbps, and several variable-rate CELP schemes have been discussed in the literature, including QCELP, the variable-rate wideband digital cellular mobile radio speech coding standard specified in IS-95. A key component of these speech coders is the detection and classification of speech activity, and several cues for rate variation have been studied, such as measuring the short-term speech energy, deciding whether the speech is voiced or unvoiced, or making more sophisticated phonetic classifications. We present a new method for rate variation based on a measure of subband spectral flatness, called spectral entropy. Spectral entropy is a normalized indicator of the texture of the input spectrum and is thus less dependent on speech and background noise energy variations. We present some results on the use of spectral entropy for voice activity detection across subbands and then evaluate using spectral entropy for deriving mode and rate allocation cues for a variable-rate CELP coder operating at an average rate of 2 kbps. To achieve communications quality speech at this rate, we develop a new split-band vector quantization (VQ) technique for representing the line spectral pairs and a multiple codebook approach for efficiently quantizing the coefficients of a three-tap pitch predictor, called lag-indexed VQ.

Channel decoding for wireless telephones with multiple microphones and multiple description transmission

Abstract: An embodiment of the present invention provides a wireless telephone including a receiver module, a channel decoder, a speech decoder, and a speaker. The receiver module receives a plurality of versions of a voice signal, wherein each version of the voice signal comprises a plurality of speech frames. The channel decoder is configured to decode a speech parameter associated with a first speech frame from a first version of the voice signal, wherein decoding the speech parameter includes selecting an optimal bit sequence from a plurality of candidate bit sequences and wherein the selection of the optimal bit sequence is based in part on a second speech frame from a second version of the voice signal. The speech decoder decodes at least one of the first or second versions of the voice signal based on the speech parameter to generate an output signal. The speaker receives the output signal and produces a sound pressure wave corresponding thereto.