• Title/Summary/Keyword: Speech processor

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Implementation of Speaker Verification Security System Using DSP Processor(TMS320C32) (DSP Processor(TMS320C32)를 이용한 화자인증 보안시스템의 구현)

  • Haam, Young-Jun;Kwon, Hyuk-Jae;Choi, Soo-Young;Jeong, lk-Joo
    • Journal of Industrial Technology
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    • v.21 no.B
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    • pp.107-116
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    • 2001
  • The speech includes various kinds of information : language information, speaker's information, affectivity, hygienic condition, utterance environment etc. when a person communicates with others. All technologies to utilize in real life processing this speech are called the speech technology. The speech technology contains speaker's information that among them and it includes a speech which is known as a speaker recognition. DTW(Dynamic Time Warping) is the speaker recognition technology that seeks the pattern of standard speech signal and the similarity degree in an inputted speech signal using dynamic programming. ln this study, using TMS320C32 DSP processor, we are to embody this DTW and to construct a security system.

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Comparison of Speech Onset Detection Characteristics of Adaptation Algorithms for Cochlear Implant Speech Processor (인공와우 어음처리방식을 위한 적응효과 알고리즘의 음성개시점 검출 특성 비교)

  • Choi, Sung-Jin;Kim, Jin-Ho;Kim, Kyung-Hwan
    • Journal of Biomedical Engineering Research
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    • v.29 no.1
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    • pp.25-31
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    • 2008
  • It is well known that temporal information, i.e speech onset, about input speech can be represented to the response nerve signal of auditory nerve better depending on the adaptation effect occurred in the auditory nerve synapse. In addition, the performance of a speech processor of cochlear implant can be improved by the adaptation effect. In this paper, we observed the emphasis characteristic of speech onset in the recently proposed adaptation algorithm, analyzed the characteristic of performance change according to the variation of parameters and compared with transient emphasis spectral maxima (TESM) is the previous typical strategy. When observing false peaks which are generated everywhere except speech onset, in the case of the proposed model, the false peak were generated much less than in the case of the TESM and it is more distinguishable under noise.

Implementation of Real-Time Adaptive Noise Cancellation System Using DSP Processor (DSP 프로세서를 이용한 실시간 ANC 시스템 구현에 관한 연구)

  • Lee Young Il;Choi Hong Sub
    • MALSORI
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    • no.52
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    • pp.121-132
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    • 2004
  • This paper is aiming at real-time implementation of adaptive noise cancellation system using DSP processor. ACHARF algorithm, which guarantees stability and fast convergence by adaptive compensator, is used on this DSP system. For the experiments, TLV320AIC23 stereo CODEC of TI Inc. is used with TMS320C6413 DSP processor. Signals of primary input and reference input are obtained by two microphones. The primary input is the voice plus noise signal and the reference input is white noise or real noise. The experimental results show that ANC system using DSP processor with ACHARF is verified to be an effective speech enhancement method for various speech processing units.

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A Simulation Study on Improvements of Speech Processing Strategy of Cochlear Implants Using Adaptation Effect of Inner Hair Cell and Auditory Nerve Synapse (청각신경 시냅스의 적응 효과를 이용한 인공와우 어음처리 알고리즘의 개선에 대한 시뮬레이션 연구)

  • Kim, Jin-Ho;Kim, Kyung-Hwan
    • Journal of Biomedical Engineering Research
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    • v.28 no.2
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    • pp.205-211
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    • 2007
  • A novel envelope extraction algorithm for speech processor of cochlear implants, called adaptation algorithm, was developed which is based on a adaptation effect of the inner hair cell(IHC)/auditory nerve(AN) synapse. We achieved acoustic simulation and hearing experiments with 12 normal hearing persons to compare this adaptation algorithm with existent standard envelope extraction method. The results shows that speech processing strategy using adaptation algorithm showed significant improvements in speech recognition rate under most channel/noise condition, compared to conventional strategy We verified that the proposed adaptation algorithm may yield better speech perception under considerable amount of noise, compared to the conventional speech processing strategy.

A Design of Multi-channel Speech Pickup Embedded System for Hands-free Comuunication (핸즈프리 통신을 위한 다중채널 음성픽업 임베디드 시스템 설계)

  • Ju, Hyng-Jun;Park, Chan-Sub;Jeon, Jae-Kuk;Kim, Ki-Man
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.11 no.2
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    • pp.366-373
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    • 2007
  • In this paper we propose a multi-channel speech pickup system for calling quality enhancement of hands-free communication using ALTERA Nios-II processor. Multi-channel speech pickup system uses Delay-and-Sum beamformer with zero-padding interpolator. This paper implements speech pickup system using the Nios-II processor with real-time I/O data processing speed. The proposes speech pickup embedded system shows a good agreement with those of computer simulation(MATLAB) and conventional DSP processor(TMS320C6711) result. The proposed method is effective more than previous methods in cost and design processing time. As a result, LE(Logic Element) of hardware used 3,649/5,980(61%) on a chip.

A Study on Design and Implementation of Embedded System for speech Recognition Process

  • Kim, Jung-Hoon;Kang, Sung-In;Ryu, Hong-Suk;Lee, Sang-Bae
    • Journal of the Korean Institute of Intelligent Systems
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    • v.14 no.2
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    • pp.201-206
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    • 2004
  • This study attempted to develop a speech recognition module applied to a wheelchair for the physically handicapped. In the proposed speech recognition module, TMS320C32 was used as a main processor and Mel-Cepstrum 12 Order was applied to the pro-processor step to increase the recognition rate in a noisy environment. DTW (Dynamic Time Warping) was used and proven to be excellent output for the speaker-dependent recognition part. In order to utilize this algorithm more effectively, the reference data was compressed to 1/12 using vector quantization so as to decrease memory. In this paper, the necessary diverse technology (End-point detection, DMA processing, etc.) was managed so as to utilize the speech recognition system in real time

An emotional speech synthesis markup language processor for multi-speaker and emotional text-to-speech applications (다음색 감정 음성합성 응용을 위한 감정 SSML 처리기)

  • Ryu, Se-Hui;Cho, Hee;Lee, Ju-Hyun;Hong, Ki-Hyung
    • The Journal of the Acoustical Society of Korea
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    • v.40 no.5
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    • pp.523-529
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    • 2021
  • In this paper, we designed and developed an Emotional Speech Synthesis Markup Language (SSML) processor. Multi-speaker emotional speech synthesis technology that can express multiple voice colors and emotional expressions have been developed, and we designed Emotional SSML by extending SSML for multiple voice colors and emotional expressions. The Emotional SSML processor has a graphic user interface and consists of following four components. First, a multi-speaker emotional text editor that can easily mark specific voice colors and emotions on desired positions. Second, an Emotional SSML document generator that creates an Emotional SSML document automatically from the result of the multi-speaker emotional text editor. Third, an Emotional SSML parser that parses the Emotional SSML document. Last, a sequencer to control a multi-speaker and emotional Text-to-Speech (TTS) engine based on the result of the Emotional SSML parser. Based on SSML which is a programming language and platform independent open standard, the Emotional SSML processor can easily integrate with various speech synthesis engines and facilitates the development of multi-speaker emotional text-to-speech applications.

Real-time Implementation of Multi-channel AMR Speech Coder (멀티채널 AMR 음성부호화기의 실시간 구현)

  • 지덕구;박만호;김형중;윤병식;최송인
    • The Journal of the Acoustical Society of Korea
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    • v.20 no.8
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    • pp.19-23
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    • 2001
  • DSP-based implementation is pervasive in wireless communication parts for systems and handsets according to developing high-speed and low-power programmable Digital Signal Processor (DSP). In this paper, we present a real-time implementation of multi-channel Adaptive Multi-rate (AMR) speech coder. The real-time implementation of an AMR algorithm is achieved using 32-bit fixed-point TMS320C6202 DSP chip that operates at 250 MHz. We performed cross compile, linear assembly optimization and TMS320C62xx assembly optimization for real-time implementation. Furthermore, speech data input/output function and communication function with external CPU is included in an AMR speech coder. The AMR Speech coder developed using DSP EVM board was evaluated in ETRI IMT-2000 Test-bed system.

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Real-time Implementation of a GSM-EFR Speech Coder on a 16 Bit Fixed-point DSP (16 비트 고정 소수점 DSP를 이용한 GSM-EFR 음성 부호화기의 실시간 구현)

  • 최민석;변경진;김경수
    • The Journal of the Acoustical Society of Korea
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    • v.19 no.7
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    • pp.42-47
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    • 2000
  • This paper describes a real-time implementation of a GSM-EFR (Global System for Mobil communications Enhanced Full Rate) speech coder using OakDSP core; a 16bit fixed-point Digital Signal Processor (DSP) by DSP Group, Inc. The real-time implemented speech coder required about 24MIPS for computation and 7.06K words and 12.19K words for code and data memory, respectively. The implemented GSM-EFR speech coder passes all of test vectors provided by ETSI (European Telecommunication Standard Institute), and perceptual speech quality measurement using MNB algorithm shows that the quality of the GSM-EFR speech coder is similar to the one of 32kbps ADPCM. The real-time implemented GSM-EFR speech coder which is the highest bit-rate mode of the GSM-AMR speech coder will be used as the basic structure of the GSM-AMR speech coder which is embedded in MODEM ASIC of IMT2000 asynchronous mode mobile station.

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Simulation of speech processing and coding strategy for cochlear implants (인공 청각 장치의 음성신호 처리와 자극방법의 시뮬레이션)

  • Kim, Young-Hoon;Park, Kwang-Suk
    • Proceedings of the KOSOMBE Conference
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    • v.1991 no.11
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    • pp.30-33
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    • 1991
  • The object of speech processor for cochlear implants is to deliver speech information to the central nerve system. In this study we have presented the method which simulate speech processing and coding strategy for cochlear implants and simulated two different processing methods to the 12 adults with normal ears. The formant sinusoidal coding was better than the formant pulse coding In the consonant perception test and learning effects.(p < 0.05)

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