Kimio Miseki

CELP speech coding based on an adaptive pulse position codebook

CELP coders using pulse codebooks for excitations such as ACELP have the advantages of low complexity and high speech quality. At low bit rates, however, the decrease of pulse position candidates and the number of pulses degrades reconstructed speech quality. This paper describes a method for adaptively allocating of pulse position candidates. In the proposed method, N efficient candidates of pulse positions are selected out of all possible positions in a subframe. The amplitude envelope of an adaptive code vector is used for selecting N efficient candidates. The larger the amplitude is, the more pulse positions are assigned. Using an adaptive code vector for the adaptation, the proposed method requires no additional bits for the adaptation. Experimental results show that the proposed method increases WSNRseg by 0.3 dB and MOS by 0.15.

Speech encoding apparatus utilizing stored code data

A learning-type speech encoding apparatus comprises an adaptive code book storing driving signal vectors, a minimum distortion searching circuit for searching the adaptive code book for an optimum driving signal vector on the basis of the input speech signal, a synthesizing filter for synthesizing a speech signal using the optimum driving signal vector retrieved, a buffer for storing the optimum driving signal vector retrieved, a training vector creating section for producing a training vector by segmenting the stored driving signal vector in units of a specified length, and a learning section for learning by constantly updating the driving signal vectors in the code book on the basis of the training vector.

CELP coding with an adaptive density pulse excitation model

An approach to dynamic bit-allocation to excitation vectors to improve the performance of a code excited linear prediction (CELP) coder is proposed. The method is based on an adaptive density pulse (ADP) excitation. By using the ADP, bit-allocation to the excitation vector can be easily varied. Also, the number of samples of excitation can be reduced. The effects of the ADP parameters on the synthetic speech quality are discussed. The ADP-CELP coder is described. The benefit of introducing the ADP excitation model to the CELP coder is evaluated. The segmental signal-to-noise ratio gains of the ADP-CELP coder over the conventional CELP are about 2 dB both at 8 kbit/s and at 4.8 kbit/s.<<ETX>>

Adaptive bit-allocation between the pole-zero synthesis filter and excitation in CELP

The authors propose a CELP (code excited linear prediction) coder with a novel adaptive bit allocation between the synthesis filter and excitation by changing the filter type between an all-pole filter and a pole-zero filter. The filter type and bit allocation are determined based on the minimum perceptually weighted error criterion. When an all-pole filter is selected as the synthesis filter, the input speech is coded in the same manner as a conventional CELP coder. When a pole-zero filter is selected, the zero filters coefficients are coded in a closed loop. Experimental results reveal that the proposed CELP coder improves segmental signal-to-noise ratio by approximately 1 dB over a conventional CELP coder at 4.3 kb/s.<<ETX>>

Improvement of ADP-CELP speech coding at 4 kbits/s

The authors describe improvements in the ADP-CELP (adaptive-density-pulse code-excited-linear-prediction) coder at 4 kb/s. A fractional delay was introduced to the adaptive codebook. A fast exhaustive search procedure for a uniformly distributed fractional delay was studied for the case where the delay length is shorter than the codevectors dimension. A novel method to determine the long-term predictors gain was also used. Four methods to find the density pattern of the ADP excitation were investigated. A modified adaptive postfilter was used to improve subjective speech quality.<<ETX>>

ARMA model based speech coding at 8 kb/s

A novel speech coding approach is proposed that gives high-quality performance at 8 kb/s. The coder is based on an autoregressive moving-average (ARMA) model as the prediction filter and a new excitation model. A simple ARMA analysis method is proposed that features a technique for eliminating the fine harmonic structure within the speech spectrum. The excitation signal is modeled as a pulse train whose density is varied depending on the residual signals power. The proposed coder produced high-quality speech comparable with 6 bit log PCM at 8 kb/s according to computer simulation.<<ETX>>