Topic
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.
Papers published on a yearly basis
Papers
More filters
TL;DR: The overall results suggest that speech intelligibility is not severely impaired as long as the filtered spectral components have a rate of change between 1 and 16 Hz.
Abstract: The intelligibility of syllables whose cepstral trajectories were temporally filtered was measured. The speech signals were transformed to their LPC cepstral coefficients, and these coefficients were passed through different filters. These filtered trajectories were recombined with the residuals and the speech signal reconstructed. The intelligibility of the reconstructed speech segments was then measured in two perceptual experiments for Japanese syllables. The effect of various low-pass, high-pass, and bandpass filtering is reported, and the results summarized using a theoretical approach based on the independence of the contributions in different modulation bands. The overall results suggest that speech intelligibility is not severely impaired as long as the filtered spectral components have a rate of change between 1 and 16 Hz.
69 citations
27 Apr 1993
TL;DR: In quantization experiments ISP has been found to compare favorably with LSP, and a study of interframe differentiation coding for ISP and LSP demonstrates the respective performances of the two sets.
Abstract: Immittance spectral pairs (ISPs) form a new set of parameters for representing the linear predictive coding (LPC) filter. For a filter of order n ISP consists of a gain and n-1 frequency parameters, instead of n frequency parameters as is the case for line spectrum pair (LSPs). In regarding LPC as a pseudo-model for the vocal tract, ISP can represent the immitance at the glottis without imposing, like LSP, artificial boundary conditions. In quantization experiments ISP has been found to compare favorably with LSP. A study of interframe differentiation coding for ISP and LSP demonstrates the respective performances of the two sets. >
69 citations
TL;DR: The speech synthesis from concept system converts an input concept into speech by using a transformational grammar to generate a well‐formed English sentence and a word concatenation synthesizer to generate the actual speech output.
Abstract: A synthesis method, called speech synthesis from concept, is described which has been designed specifically for providing speech output from information systems. It differs from conventional techniques in that data is passed from the information system to the speech synthesis system, not in the form of text or phonetic transcription, but in the form of an abstract structure called an input concept. The speech synthesis from concept system converts an input concept into speech by using a transformational grammar to generate a well‐formed English sentence and a word concatenation synthesizer to generate the actual speech output. The ’’top down’’ nature of this process reduces the computation required within the information system and enables high‐quality speech to be produced.
69 citations
17 Sep 2008
TL;DR: In this paper, the state of the art in important areas of speech and audio signal processing is discussed, including multi-microphone systems, specific approaches for noise reduction, and evaluations of speech signals and speech processing systems.
Abstract: The book reflects the state of the art in important areas of speech and audio signal processing. It presents topics which are missed so far and most recent findings in the field. Leading international experts report on their field of work and their new results. Considerable amount of space is covered by multi-microphone systems, specific approaches for noise reduction, and evaluations of speech signals and speech processing systems. Multi-microphone systems include automatic calibration of microphones, localisation of sound sources, and source separation procedures. Also covered are recent approaches to the problem of adaptive echo and noise suppression. A novel solution allows the design of filter banks exhibiting bands spaced according to the Bark scale und especially short delay times. Furthermore, a method for engine noise reduction and proposals for improving the signal/noise ratio based on partial signal reconstruction or using a noise reference are reported. A number of contributions deal with speech quality. Besides basic considerations for quality evaluation specific methods for bandwidth extension of telephone speech are described. Procedures to reduce the reverberation of audio signals can help to increase speech intelligibility and speech recognition rates. In addition, solutions for specific applications in speech and audio signal processing are reported including, e.g., the enhancement of audio signal reproduction in automobiles and the automatic evaluation of hands-free systems and hearing aids.
68 citations
01 Apr 1981
TL;DR: An LPC base-band vocoder is developed and experiments have shown the coder to be robust to background noise and implementation aspects as well as simulation results are discussed.
Abstract: An LPC base-band vocoder is developed. The novel feature concerns the coding of the base-band. A model is set up for the base-band as a set of modulated tones. Algorithms are presented for the extraction of amplitude and phase/frequency of the tones. Implementation aspects as well as simulation results are discussed. Total bit rates in the order of 3,2-4.8 kbits are possible where approximately one half of the bits represents the base-band coding. Experiments have shown the coder to be robust to background noise.
67 citations