Makoto Morito
Oki Electric Industry
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Publication
Featured researches published by Makoto Morito.
Journal of the Acoustical Society of America | 1992
Takashi Yazu; Makoto Morito
In a sub-band speech analyzing and synthesizing apparatus, a low-pass filter comprises a nonrecursive filter. The center frequency and bandwidth of each channel are selected so that the decimated sampling period of that channel is an integer multiple of the period of the modulating signal or the demodulating signal of the channel. Modulation or demodulation is performed simultaneously with the low-pass filtering by the nonrecursive filter.
Journal of the Acoustical Society of America | 1989
Makoto Morito
A speech synthesizer for reproducing natural speech with small memory. Synthesized speech consists of symmetric speech segments. Only half of a symmetric speech segment is stored in a memory through ADPCM code, and the other half of the segment is generated by using the stored half segment. Repeat control data is also stored in a memory so that only a single segment is enough to be stored when the same segments are repeated. In the repeated process, the amplitude of the segment is changed smoothly so that amplitude discontinuity between the two successive segments does not occur. The synthesized digital speech data is converted to an analog waveform by using a digital to analog converter.
Journal of the Acoustical Society of America | 1989
Takashi Yazu; Makoto Morito
Most sub‐band coders (SBC) have been realized by quadrature‐mirror filter (QMF) with frequency shift and low‐pass filter (LPF). However, the QMF method has the following problems: (i) Band separation is limited to the power of 2; (ii) arithmetic errors are accumulated because of the cascade connection of QMF; and (iii) it is necessary to adjust the time delay if the number of QMF connections is different. This paper presents a new architecture for SBC with flexible band assignment, less arithmetic calculations, and small storage memory. In this SBC architecture, a finite impulse response filter (FIRF) is used instead of the QMF. The center frequency of each band is selected according to the equation Fs/Fk = D *n/m, where Fs is the sampling frequency of the input signal, Fk is the center frequency of the k th band, D is the decimation factor, and m, n are integers. According to this relationship, frequency shift and the LPF can be performed at the same time. In this case, data memory for the FIRF is drasti...
Journal of the Acoustical Society of America | 1991
Makoto Morito; Yukio Tabei; Kozo Yamada
Archive | 1989
Makoto Morito; Yukio Tabei; Kozo Yamada
Archive | 2010
Makoto Morito; Takashi Yazu; Kei Yamada; Tetsunori Kobayashi; Kenzo Akagiri; Tetsuji Ogawa
Archive | 1985
Makoto Morito; Masao Takeuchi; Akihiko Fujisawa; Yukio Tabei; Keiko Takahashi
conference of the international speech communication association | 1990
Hiroki Kamanaka; Takashi Yazu; Keiichi Chihara; Makoto Morito
Archive | 2007
Makoto Morito
Archive | 2009
Kenzo Akagiri; Tetsunori Kobayashi; Makoto Morito; Tetsuji Ogawa; Kei Yamada; Takashi Yato; 哲司 小川; 哲則 小林; 圭 山田; 誠 森戸; 隆 矢頭; 健三 赤桐