• Title/Summary/Keyword: speech source

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A Study on Multi-Pulse Speech Coding Method by Using V/S/TSIUVC (V/S/TSIUVC를 이용한 멀티펄스 음성부호화 방식에 관한 연구)

  • Lee See-Woo
    • Journal of Korea Multimedia Society
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    • v.7 no.9
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    • pp.1233-1239
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    • 2004
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. This paper present a new multi-pulse coding method by using V/S/TSIUVC switching, individual pitch pulses and TSIUVC approximation-synthesis method in order to restrict a distortion of speech quality. The TSIUVC is extracted by using the zero crossing rate and individual pitch pulse. And the TSIUVC extraction rate was 91% for female voice and 96.2% for male voice respectively. The important thing is that the frequency information of 0.347kHz below and 2.813kHz above can be made with high quality synthesis waveform within TSIUVC. I evaluate the MPC use V/UV and the FBD-MPC use V/S/TSIUVC. As a result, I knew that synthesis speech of the FBD-MPC was better in speech quality than synthesis speech of the MPC.

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Multi-channel input-based non-stationary noise cenceller for mobile devices (이동형 단말기를 위한 다채널 입력 기반 비정상성 잡음 제거기)

  • Jeong, Sang-Bae;Lee, Sung-Doke
    • Journal of the Korean Institute of Intelligent Systems
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    • v.17 no.7
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    • pp.945-951
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    • 2007
  • Noise cancellation is essential for the devices which use speech as an interface. In real environments, speech quality and recognition rates are degraded by the auditive noises coming near the microphone. In this paper, we propose a noise cancellation algorithm using stereo microphones basically. The advantage of the use of multiple microphones is that the direction information of the target source could be applied. The proposed noise canceller is based on the Wiener filter. To estimate the filter, noise and target speech frequency responses should be known and they are estimated by the spectral classification in the frequency domain. The performance of the proposed algorithm is compared with that of the well-known Frost algorithm and the generalized sidelobe canceller (GSC) with an adaptation mode controller (AMC). As performance measures, the perceptual evaluation of speech quality (PESQ), which is the most widely used among various objective speech quality methods, and speech recognition rates are adopted.

A Study on TSIUVC Approximate-Synthesis Method using Least Mean Square and Frequency Division (주파수 분할 및 최소 자승법을 이용한 TSIUVC 근사합성법에 관한 연구)

  • 이시우
    • Journal of Korea Multimedia Society
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    • v.6 no.3
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    • pp.462-468
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    • 2003
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech quality in case coexist with a voiced and an unvoiced consonants in a frame. So, I propose TSIUVC(Transition Segment Including Unvoiced Consonant) searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This paper present a new method of TSIUVC approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TSIUVC by using frequency information of 0.547KHz below and 2.813KHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TSIUVC. This method has the capability of being applied to a new speech coding of Voiced/Silence/TSIUVC, speech analysis and speech synthesis.

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A New Endpoint Detection Method Based on Chaotic System Features for Digital Isolated Word Recognition System

  • Zang, Xian;Chong, Kil-To
    • Proceedings of the IEEK Conference
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    • 2009.05a
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    • pp.37-39
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    • 2009
  • In the research of speech recognition, locating the beginning and end of a speech utterance in a background of noise is of great importance. Since the background noise presenting to record will introduce disturbance while we just want to get the stationary parameters to represent the corresponding speech section, in particular, a major source of error in automatic recognition system of isolated words is the inaccurate detection of beginning and ending boundaries of test and reference templates, thus we must find potent method to remove the unnecessary regions of a speech signal. The conventional methods for speech endpoint detection are based on two simple time-domain measurements - short-time energy, and short-time zero-crossing rate, which couldn't guarantee the precise results if in the low signal-to-noise ratio environments. This paper proposes a novel approach that finds the Lyapunov exponent of time-domain waveform. This proposed method has no use for obtaining the frequency-domain parameters for endpoint detection process, e.g. Mel-Scale Features, which have been introduced in other paper. Comparing with the conventional methods based on short-time energy and short-time zero-crossing rate, the novel approach based on time-domain Lyapunov Exponents(LEs) is low complexity and suitable for Digital Isolated Word Recognition System.

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Development of 3-Ch EGG System Using Modulation and Demodulation Techniques(I) (변복조 방식을 이용한 3-채널 EGG 시스템의 개발(I))

  • Kim, J.M.;Song, C.G.;Lee, M.H.
    • Proceedings of the KOSOMBE Conference
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    • v.1993 no.05
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    • pp.134-135
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    • 1993
  • The purpose of this research is development of EGG system for quantitative assessment of laryngeal function using speech and electroglotto-graphic data. The designed EGG system is 4-electrodes system which excitation current source is supplied from 1st to 4th electrode. The output signal.: from 2nd and 3rd electrodes, which are motivated by frequency of excitation current source, are air-pressure waveforms from vocal folds. After demodulation process, we obtain pitch signals of the modulated waveforms by excitation current source through differentiator which cuts off frequency below 0.1Hz. Software processing methods were used as conventional pitch extraction methods, but the proposed system is designed to analog hardware in order to eliminate interferences from low formant frequency of speech. We will construct the discriminating database between pathological subjects and control groups on each case. Using the proposed 3 channel EGG system and LMS algorithm, it will be detected that the distinctive characteristics of laryngeal function of voiced region and other regions by EGG signals and LPC spectra.

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Speech Transition Detection and approximate-synthesis Method for Speech Signal Compression and Recovery (음성신호 압축 및 복원을 위한 음성 천이구간 검출과 근사합성 방식)

  • Lee, Kwang-Seok;Kim, Bong-Gi;Kang, Seong-Soo;Kim, Hyun-Deok
    • Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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    • 2008.05a
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    • pp.763-767
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    • 2008
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high quality approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.

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Design of a Low Bit-rate Speech Coder Based on Mixed Multi-band Excitation Model (혼합 다중대역 여기모델에 기반한 저 전송률 음성 부호화기의 설계)

  • 한우진;오영환
    • The Journal of the Acoustical Society of Korea
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    • v.21 no.6
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    • pp.510-521
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    • 2002
  • MBE (multi-band excitation) coder can achieve high qualify synthetic speech below 4.0 kbps. There are, however, significant differences of the fine structure between the original spectrum and the synthetic spectrum. They are mainly due to the exclusive partition of voiced and unvoiced regions in frequency domain and the decision procedure based on the experimental threshold. This paper proposes MMBE (mixed multi-band excitation) speech model to overcome drawbacks of a MBE coder. In addition, two analysis methods, which do not need my decision procedure based on a threshold, are presented. Both voiced and unvoiced components can be mixed over all the frequency axis in the MMBE speech model. To illustrate the potential of the proposed speech model, we develop a 2.6 kbps MMBE coder and compare it with a 2.9 kbps MBE coder by both objective and subjective methods. The results have shown that the proposed coder has a better performance even at a lower bit-rate compared with the MBE coder.

Speech Signal Compression and Recovery Using Transition Detection and Approximate-Synthesis (천이구간 추출 및 근사합성에 의한 음성신호 압축과 복원)

  • Lee, Kwang-Seok;Lee, Byeong-Ro
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.13 no.2
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    • pp.413-418
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    • 2009
  • In a speech coding system using excitation source of voiced and unvoiced, it would be involved a distortion of speech qualify in case coexist with a voiced and an unvoiced consonants in a frame. So, We proposed TS(Transition Segment) including unvoiced consonant searching and extraction method in order to uncoexistent with a voiced and unvoiced consonants in a frame. This research present a new method of TS approximate-synthesis by using Least Mean Square and frequency band division. As a result, this method obtain a high qualify approximation-synthesis waveforms within TS by using frequency information of 0.547kHz below and 2.813kHz above. The important thing is that the maximum error signal can be made with low distortion approximation-synthesis waveform within TS. This method has the capability of being applied to a new speech coding of Voiced/Silence/TS, speech analysis and speech synthesis.

Combining multi-task autoencoder with Wasserstein generative adversarial networks for improving speech recognition performance (음성인식 성능 개선을 위한 다중작업 오토인코더와 와설스타인식 생성적 적대 신경망의 결합)

  • Kao, Chao Yuan;Ko, Hanseok
    • The Journal of the Acoustical Society of Korea
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    • v.38 no.6
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    • pp.670-677
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    • 2019
  • As the presence of background noise in acoustic signal degrades the performance of speech or acoustic event recognition, it is still challenging to extract noise-robust acoustic features from noisy signal. In this paper, we propose a combined structure of Wasserstein Generative Adversarial Network (WGAN) and MultiTask AutoEncoder (MTAE) as deep learning architecture that integrates the strength of MTAE and WGAN respectively such that it estimates not only noise but also speech features from noisy acoustic source. The proposed MTAE-WGAN structure is used to estimate speech signal and the residual noise by employing a gradient penalty and a weight initialization method for Leaky Rectified Linear Unit (LReLU) and Parametric ReLU (PReLU). The proposed MTAE-WGAN structure with the adopted gradient penalty loss function enhances the speech features and subsequently achieve substantial Phoneme Error Rate (PER) improvements over the stand-alone Deep Denoising Autoencoder (DDAE), MTAE, Redundant Convolutional Encoder-Decoder (R-CED) and Recurrent MTAE (RMTAE) models for robust speech recognition.

A study on Speech Coding Method using V/S/TSIUVC Switching (V/S/TSIUVC 스위칭을 이용한 음성부호화 방식에 관한 연구)

  • Lee, See-Woo
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.7 no.6
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    • pp.1180-1184
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    • 2006
  • In a speech coding system using excitation source of voiced and unvoiced, it would be a distortion of speech quality in a voiced and an unvoiced consonants in a frame. In this paper, I propose a new multi-pulse coding method make use of V/S/TSIUVC switching and TSIUVC approximation-synthesis method in order to restrict a distortion of speech quality. The TSIUVC is extracted by using the zero crossing rate and individual pitch pulse. And the TSIUVC extraction rate was 91% for female voice and 96.2% for male voice. The important thing is that the frequency information of 0.547kHz below and 2.813kHz above can be made with high quality synthesis waveform within TSIUVC. I evaluated the MPC of V/UV and FBD-MPC of V/S/TSIUVC. As a result, the synthesis speech of FBD-MPC was better in speech quality than the MPC.

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