• Title/Summary/Keyword: Codebook-based Speech Enhancement

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Gain Compensation Method for Codebook-Based Speech Enhancement (코드북 기반 음성향상 기법을 위한 게인 보상 방법)

  • Jung, Seungmo;Kim, Moo Young
    • Journal of the Institute of Electronics and Information Engineers
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    • v.51 no.9
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    • pp.165-170
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    • 2014
  • Speech enhancement techniques that remove surrounding noise are stressed to preprocessor of speech recognition. Among the various speech enhancement techniques, Codebook-based Speech Enhancement (CBSE) operates efficiently in non-stationary noise environments. But, CBSE has some problems that inaccurate gains can be estimated if mismatch occur between input noisy signal and trained speech/noise codevectors. In this paper, the Normalized Weighting Factor (NWF) is calculated by long-term noise estimation algorithm based on Signal-to-Noise Ratio, compensated to the conventional inaccurate gains. The proposed CBSE shows better performance than conventional CBSE.

A Selection Method of Reliable Codevectors using Noise Estimation Algorithm (잡음 추정 알고리즘을 이용한 신뢰성 있는 코드벡터 조합의 선정 방법)

  • Jung, Seungmo;Kim, Moo Young
    • Journal of the Institute of Electronics and Information Engineers
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    • v.52 no.7
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    • pp.119-124
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    • 2015
  • Speech enhancement has been required as a preprocessor for a noise robust speech recognition system. Codebook-based Speech Enhancement (CBSE) is highly robust in nonstationary noise environments compared with conventional noise estimation algorithms. However, its performance is severely degraded for the codevector combinations that have lower correlation with the input signal since CBSE depends on the trained codebook information. To overcome this problem, only the reliable codevector combinations are selected to be used to remove the codevector combinations that have lower correlation with input signal. The proposed method produces the improved performance compared to the conventional CBSE in terms of Log-Spectral Distortion (LSD) and Perceptual Evaluation of Speech Quality (PESQ).

Non-Stationary/Mixed Noise Estimation Algorithm Based on Minimum Statistics and Codebook Driven Short-Term Predictor Parameter Estimation (최소 통계법과 Short-Term 예측계수 코드북을 이용한 Non-Stationary/Mixed 배경잡음 추정 기법)

  • Lee, Myeong-Seok;Noh, Myung-Hoon;Park, Sung-Joo;Lee, Seok-Pil;Kim, Moo-Young
    • The Journal of the Acoustical Society of Korea
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    • v.29 no.3
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    • pp.200-208
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    • 2010
  • In this work, the minimum statistics (MS) algorithm is combined with the codebook driven short-term predictor parameter estimation (CDSTP) to design a speech enhancement algorithm that is robust against various background noise environments. The MS algorithm functions well for the stationary noise but relatively not for the non-stationary noise. The CDSTP works efficiently for the non-stationary noise, but not for the noise that was not considered in the training stage. Thus, we propose to combine CDSTP and MS. Compared with the single use of MS and CDSTP, the proposed method produces better perceptual evaluation of speech quality (PESQ) score, and especially works excellent for the mixed background noise between stationary and non-stationary noises.

Developing a Low Power BWE Technique Based on the AMR Coder (AMR 기반 저 전력 인공 대역 확장 기술 개발)

  • Koo, Bon-Kang;Park, Hee-Wan;Ju, Yeon-Jae;Kang, Sang-Won
    • The Journal of the Acoustical Society of Korea
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    • v.30 no.4
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    • pp.190-196
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    • 2011
  • Bandwidth extension is a technique to improve speech quality and intelligibility, extending from 300-3400 Hz narrowband speech to 50-7000 Hz wideband speech. This paper designs an artificial bandwidth extension (ABE) module embedded in the AMR (adaptive multi-rate) decoder, reducing LPC/LSP analysis and algorithm delay of the ABE module. We also introduce a fast search codebook mapping method for ABE, and design a low power BWE technique based on the AMR decoder. The proposed ABE method reduces the computational complexity and the algorithm delay, respectively, by 28 % and 20 msec, compared to the traditional DTE (decode then extend) method. We also introduce a weighted classified codebook mapping method for constructing the spectral envelope of the wideband speech signal.

Artificial speech bandwidth extension technique based on opus codec using deep belief network (심층 신뢰 신경망을 이용한 오푸스 코덱 기반 인공 음성 대역 확장 기술)

  • Choi, Yoonsang;Li, Yaxing;Kang, Sangwon
    • The Journal of the Acoustical Society of Korea
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    • v.36 no.1
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    • pp.70-77
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    • 2017
  • Bandwidth extension is a technique to improve speech quality, intelligibility and naturalness, extending from the 300 ~ 3,400 Hz narrowband speech to the 50 ~ 7,000 Hz wideband speech. In this paper, an Artificial Bandwidth Extension (ABE) module embedded in the Opus audio decoder is designed using the information of narrowband speech to reduce the computational complexity of LPC (Linear Prediction Coding) and LSF (Line Spectral Frequencies) analysis and the algorithm delay of the ABE module. We proposed a spectral envelope extension method using DBN (Deep Belief Network), one of deep learning techniques, and the proposed scheme produces better extended spectrum than the traditional codebook mapping method.