Author Search Result

[Author] Miki SATO(3hit)

1-3hit
  • An Adaptive Noise Canceller with Low Signal-Distortion Based on Variable Stepsize Subfilters for Human-Robot Communication

    Miki SATO  Akihiko SUGIYAMA  Shin'ichi OHNAKA  

     
    PAPER-Digital Signal Processing

      Vol:
    E88-A No:8
      Page(s):
    2055-2061

    This paper proposes an adaptive noise canceller (ANC) with low signal-distortion for human-robot communication. The proposed ANC has two sets of adaptive filters for noise and crosstalk; namely, main filters (MFs) and subfilters (SFs) connected in parallel thereto. To reduce signal-distortion in the output, the stepsizes for coefficient adaptation in the MFs are controlled according to estimated signal-to-noise ratios (SNRs) of the input signals. This SNR estimation is carried out using SF output signals. The stepsizes in the SFs are determined based on the ratio of the primary and the reference input signals to cope with a wider range of SNRs. This ratio is used as a rough estimate of the input signal SNR at the primary input. Computer simulation results using TV sound and human voice recorded in a carpeted room show that the proposed ANC reduces both residual noise and signal-distortion by as much as 20 dB compared to the conventional ANC. Evaluation in speech recognition with this ANC reveals that with a realistic TV sound level, as good recognition rate as in the noise-free condition is achieved.

  • Near-Field Sound-Source Localization Based on a Signed Binary Code

    Miki SATO  Akihiko SUGIYAMA  Osamu HOSHUYAMA  Nobuyuki YAMASHITA  Yoshihiro FUJITA  

     
    PAPER-Digital Signal Processing

      Vol:
    E88-A No:8
      Page(s):
    2078-2086

    This paper proposes near-field sound-source localization based on crosscorrelation of a signed binary code. The signed binary code eliminates multibit signal processing for simpler implementation. Explicit formulae with near-field assumption are derived for a two microphone scenario and extended to a three microphone case with front-rear discrimination. Adaptive threshold for enabling and disabling source localization is developed for robustness in noisy environment. The proposed sound-source localization algorithm is implemented on a fixed-point DSP. Evaluation results in a robot scenario demonstrate that near-field assumption and front-rear discrimination provides almost 40% improvement in DOA estimation. A correct detection rate of 85% is obtained by a robot in a home environment.

  • A Single-Chip Speech Dialogue Module and Its Evaluation on a Personal Robot, PaPeRo-Mini

    Miki SATO  Toru IWASAWA  Akihiko SUGIYAMA  Toshihiro NISHIZAWA  Yosuke TAKANO  

     
    PAPER-Digital Signal Processing

      Vol:
    E93-A No:1
      Page(s):
    261-271

    This paper presents a single-chip speech dialogue module and its evaluation on a personal robot. This module is implemented on an application processor that was developed primarily for mobile phones to provide a compact size, low power-consumption, and low cost. It performs speech recognition with preprocessing functions such as direction-of-arrival (DOA) estimation, noise cancellation, beamforming with an array of microphones, and echo cancellation. Text-to-speech (TTS) conversion is also equipped with. Evaluation results obtained on a new personal robot, PaPeRo-mini, which is a scale-down version of PaPeRo, demonstrate an 85% correct rate in DOA estimation, and as much as 54% and 30% higher speech recognition rates in noisy environments and during robot utterances, respectively. These results are shown to be comparable to those obtained by PaPeRo.

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