Codebook

About: Codebook is a research topic. Over the lifetime, 8492 publications have been published within this topic receiving 115995 citations.

Speech codec employing speech classification for noise compensation

Abstract: A method of encoding an input speech signal using a multi-rate encoder having a plurality of encoding rates is disclosed. A high-pass filter and then a perceptual weighting filter are applied to such signal to generate a first target signal. An adaptive codebook vector is identified from an adaptive codebook using the first target signal by filtering the vector to generate a filtered adaptive codebook vector. An adaptive codebook gain for the adaptive codebook vector is calculated and an error signal minimized. The adaptive codebook gain is adaptively reduced based on one encoding rate from the plurality of encoding rates to generate a reduced adaptive codebook gain. A second target signal based at least on the first target signal and the reduced adaptive codebook gain is generated. The input speech signal is converted into an encoded speech based on the second target signal.

On reducing computational complexity of codebook search in CELP coder through the use of algebraic codes

Abstract: A general framework is introduced which allows both fast search and freedom in designing codebooks with good statistical properties. Several previously proposed schemes are compared from this viewpoint. A backward filtering formulation is given to show that sparse algebraic codes (SACs) (i.e., with few nonzero components) offer distinct advantages. It is shown that they reduce the optimal-search computation per codeword. They also allow control of the statistical properties of the codebook in the time and frequency domains. This control can be dynamic in the sense that it can be made to evolve as a function of the linear predictive coding model A(z). The algebraic-code excited linear prediction (ACELP) technology which allows full duplex operation on a single TMS320C25 at rates between 4.8 and 16 kb/s and which is based on SAC-driven dynamic codebooks is described. >

Beamforming Codebook Design and Performance Evaluation for 60GHz Wideband WPANs

Abstract: The paper proposes a codebook design to support beamforming mechanism in a wide band millimeter-wave 60GHz wireless personal area networks (60GHz WPANs) in a realistic millimeter-wave environment. The codebooks are designed sym- metrically in order to mitigate the possible beam shift due to the large differences of wave lengthes at different sub-bands of a wide band communication system. In order to provide a high data rate and high performance with minimum power consumption in 60GHz systems, the codebooks are generated with 90 degree phase resolution without amplitude adjustment. The codebooks are designed for different numbers of antenna elements, supporting a multitude of antenna configurations. The paper provides codebook design mechanism, some example codebooks and some analysis on antenna gain loss due to phase shift errors of phase shifters. Simulation results shows that: (1) To keep the gain loss at the intersections of any two patterns inside the codebook lower than 1dB, the number of patterns inside the codebook shall be at least 2 times of the number of antenna elements used for pattern generation; (2) The designed codebooks are robust to phase shift errors: the gain loss is lower than 1dB with only 10% outage probability when standard deviation of phase shift errors at phase shifters is 0.5 (28.6 degree). The proposed codebooks are a feasible design, they are the key basis of the beamforming protocol in 60GHz WPANs (IEEE 802.15.3c).

Wear-toll quality 4.8 kbps speech codec

Abstract: A speech codec operating at low data rates uses an iterative method to jointly optimize pitch and gain parameter sets. A 26-bit spectrum filter coding scheme may be used, involving successive subtractions and quantizations. The codec may preferably use a decomposed multipulse excitation model, wherein the multipulse vectors used as the excitation signal are decomposed into position and amplitude codewords. Multipulse vectors are coded by comparing each vector to a reference multipulse vector and quantizing the resulting difference vector. An expanded multipulse excitation codebook and associated fast search method, optionally with a dynamically-weighted distortion measure, allow selection of the best excitation vector without memory or computational overload. In a dynamic bit allocation technique, the number of bits allocated to the pitch and excitation signals depend on whether the signals are "significant" or "insignificant". Silence/speech detection is based on an average signal energy over an interval and a minimum average energy over a predetermined number of intervals. Adaptive post-filter and the automatic gain control schemes are also provided. Interpolation is used for spectrum filter smoothing, and an algorithm is provided for ensuring stability of the spectrum filter. Specially designed scalar quantizers are provided for the pitch gain and excitation gain.

Speech enhancement using a mixture-maximum model.

Abstract: We present a spectral domain, speech enhancement algorithm. The new algorithm is based on a mixture model for the short time spectrum of the clean speech signal, and on a maximum assumption in the production of the noisy speech spectrum. In the past this model was used in the context of noise robust speech recognition. In this paper we show that this model is also effective for improving the quality of speech signals corrupted by additive noise. The computational requirements of the algorithm can be significantly reduced, essentially without paying performance penalties, by incorporating a dual codebook scheme with tied variances. Experiments, using recorded speech signals and actual noise sources, show that in spite of its low computational requirements, the algorithm shows improved performance compared to alternative speech enhancement algorithms.