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检索条件"主题词=Linear predictive coding"

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MELP: the new Federal Standard at 2400 bps

MELP: the new Federal Standard at 2400 bps

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： L.M. Supplee R.P. Cohn J.S. Collura A.V. McCree U.S. Department of Defense Fort Meade MD USA Corporate Research & Development Texas Instrumenits Inc. Dallas TX USA

This paper describes the new U.S. Federal Standard at 2400 bps. The mixed excitation linear prediction (MELP) coder was chosen by the DoD Digital Voice Processing Consortium to replace the existing 2400 bps Federal Standard FS 1015 (LPC-10). This new standard provides equal or improved performance over the 4800 bps Federal Standard FS 1016 (CELP) at a rate equivalent to LPC-10. The MELP coder is based on the traditional LPC model, but includes additional features to improve its performance.

关键词： Band pass filters Standards development Code standards Instruments linear predictive coding Testing Communication standards Military standards Signal processing algorithms Matched filters

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Speaker verification based on a vector quantization approach that incorporates speaker cohort models and a linear discriminator

Speaker verification based on a vector quantization approach...

引用

IEEE International Conference on Systems, Man and Cybernetics

作者： W.R. Belfield R.P. Mikkilineni Lucent Technologies Inc. Columbus OH USA

This report describes a speaker verification system based on a vector quantization (VQ) approach that incorporates dynamic time warping (DTW), cohort models, and a discriminator to separate the true speakers and imposters. This system is designed for telephone network applications and provides good performance for different telephone handsets and network conditions. The speaker verification system is text dependent, and each subscriber uses a personal pass phrase to verify his/her identity. A distortion vector is computed using VQ encoding and DTW distortions with respect to speaker-dependent and cohort code books. The components of this distortion vector are applied to a linear discriminator to validate the identity of the speaker. A speaker verification database was used to evaluate the performance of this approach, and it was found that the performance is significantly better compared with that of a basic VQ system. The system achieved an equal error rate (EER) of 0.92% when the true speaker and the imposters spoke different pass phrases and 4.30% when they spoke the same phrase.

关键词： Vector quantization Cepstral analysis linear predictive coding Speech Telephony Encoding Testing Telephone sets Books Error analysis

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The multimodal multipulse excitation vocoder

The multimodal multipulse excitation vocoder

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： T. Unno T.P. Barnwell M.A. Clements LSI laboratories Asahi Chemical Industry Astugi Kanagawa Japan School of Electrical and Computer Engineering Georgia Institute of Technology Atlanta GA USA

This paper presents a new high-quality, variable-rate vocoder in which the average bit-rate is parametrically controllable. The new vocoder is intended for use with data-voice simultaneous channel (DVSC) applications, in which the speech data is transmitted simultaneously with video and other types of data. The vocoder presented in this paper achieves state-of-the-art quality at several different bit-rates between 5.5 kbps and 10 kbps. Further, it achieves this performance at acceptable levels of complexity and delay.

关键词： Vocoders Delay Speech analysis linear predictive coding Large scale integration Laboratories Chemical industry Chemical technology Electric variables control Application software

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Efficient mixed excitation models in LPC based prototype interpolation speech coders

Efficient mixed excitation models in LPC based prototype int...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： C. Papanastasiou C.S. Xydeas Speech Processing Research Laboratory Electrical Engineering Division Manchester School of Engineering University of Manchester Institute of Science and Technology Manchester UK

This paper presents a new and efficient method for modeling voiced, mixed excitation spectra in sinusoidal (SC) and prototype interpolation coding (PIC) systems. Speech harmonics are classified as "weak-voiced" or "strong-voiced" by simply examining the short-term residual magnitude spectrum. This information is encoded effectively in terms of fixed width frequency bands and is used to control sets of periodic and random sine wave oscillators which model the short-term mixed excitation nature of speech. In this way the model allows the mixing of periodic and random signal energy on a harmonic basis. The proposed methodology has been used in a 2.4 Kbits/sec speech coder, whose recovered speech quality is better than that of the 4.8 Kbits/sec DoD standard.

关键词： linear predictive coding Prototypes Interpolation Speech synthesis Equations Speech processing Signal synthesis Frequency Signal processing Speech codecs

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A 2000 bps LPC vocoder based on multiband excitation

A 2000 bps LPC vocoder based on multiband excitation

引用

IEEE International Symposium on Circuits and Systems (ISCAS)

作者： J.S. Mao Z. Chen S.C. Chan T.S. Ng K.L. Ho Department of Electrical and Electronic Engineering University of Hong Kong Hong Kong China

This paper presents an improved mixed LPC vocoder at 2000 bps using Multi-Band Excitation analysis by a synthesis algorithm. The new vocoder determines the voiced/unvoiced characteristics harmonic by harmonic in a frame, and finds the first voiced/unvoiced transition as the cut-off frequency, which is more accurate and efficient than traditional cut-off frequency detection. The synthetic speech below the cut-off frequency is excited by a series of voiced harmonics, while the signal above the cut-off frequency is simulated by a noise source. The final output speech is the sum of these two outputs. To increase the naturalness and clearness of the synthesized speech, this model applies phase prediction and spectral enhancement in the synthesizer. It is also possible to reduce the bit rate to 1200 bps. Informal listening tests indicate that the output speech possesses higher intelligibility and quality than that of the 2.4 kbps LPC-10e standard, and is comparable with the 4.8 kbps FS1016 CELP vocoder.

关键词： linear predictive coding Vocoders Cutoff frequency Speech synthesis Speech enhancement Algorithm design and analysis Signal synthesis predictive models Synthesizers Bit rate

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Toll quality variable-rate speech codec

Toll quality variable-rate speech codec

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： P. Ojala Speech and Audio Systems Laboratory Nokia Research Center Tampere Finland

This paper presents a source controlled variable-rate CELP type speech codec. First, a voice activity detection block distinguishes active speech frames from silence and background noise. The active speech is further classified into voiced and unvoiced frames. The voiced frames have variable bit-rate pitch-lag quantization based on the characteristics of the speech, whereas the unvoiced frames are coded without pitch information. A variable bit-rate fixed codebook excitation with a variable number of excitation pulses is determined for each speech frame. The performance of the linear analysis part of the codec as well as the input speech characteristics determine the excitation bit-rate. The average bit-rate of the codec is around 7.0 kbit/s for active speech, and the overall bit-rate ranges from 0 to 7.85 kbit/s. The described variable-rate codec produces toll quality speech equal to that of the 32 kbit/s ADPCM (G.726) standard.

关键词： Speech codecs Speech analysis Quantization Speech coding Speech enhancement Performance analysis Decoding Signal analysis Speech synthesis linear predictive coding

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A robust 1200 bps voice coder based on vector quantization of line spectral frequencies

A robust 1200 bps voice coder based on vector quantization o...

引用

MILCOM, Military Communications Conference

作者： Shan Lu M. Baraniecki COMSAT Laboratories Clarksburg MD USA

This paper describes a 1200 bps voice codec that is based on a 10th-order linear prediction analysis, split vector quantization of line spectral frequencies, and differential pitch quantization. Robust codebooks and error protection and concealment techniques are used to minimize the effect of channel errors. Listening tests show that the codec has a high degree of speech intelligibility and speaker recognizability with natural voice quality.

关键词： Robustness Vector quantization Frequency Books Speech coding Speech analysis Protection Speech codecs Bandwidth linear predictive coding

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A comparison of the new 2400 bps MELP Federal Standard with other standard coders

A comparison of the new 2400 bps MELP Federal Standard with ...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： M.A. Kohler U.S. Department of Defense Fort Meade MD USA

In 1996, the U.S. Department of Defense Digital Voice Processing Consortium (DDVPC) selected Texas Instrument's mixed excitation linear prediction (MELP) algorithm as the recommended new Federal Standard for 2400 bps voice communications. The algorithm selection process involved quality, intelligibility, communicability, and recognizability testing in many acoustic noise, error, and tandem conditions. Algorithm complexity was also measured. This paper compares the performance scores, diagnostic information, and complexity of MELP to the 4800 bps Federal Standard (FS1016) code excited linear prediction (CELP) algorithm, the 16 kbps continuously variable slope delta modulation (CVSD) algorithm, and the venerable Federal Standard (FIPS Pub. 137) 2400 bps linear predictive coding (LPC-10) algorithm.

关键词： Code standards Acoustic noise Prediction algorithms Acoustic testing linear predictive coding Delta modulation Working environment noise Microphones Instruments Communication standards

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Variable time-scale modification of speech using transient information

Variable time-scale modification of speech using transient i...

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： Sungjoo Lee Hee Dong Kim Hyung Soon Kim Department of Electronics Engineering Pusan National University South Korea Department of Telematics Engineering University of Suwon South Korea

ISBN: (纸本)0818679190

Conventional time-scale modification methods have the problem that as the modification rate gets higher the time-scale modified speech signal becomes less intelligible, because they ignore the effect of articulation rate on speech characteristics. We propose a variable time-scale modification method based on the knowledge that the timing information of transient portions of a speech signal plays an important role in speech perception. After identifying transient and steady portions of a speech signal, the proposed method gets the target rate by modifying steady portions only. The result of subjective preference test indicates that the proposed method produces performance superior to that of the conventional SOLA method.

关键词： Timing Testing Speech coding Telematics Signal processing Bit rate Telephony Natural languages linear predictive coding Cepstral analysis

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Speech enhancement using CSS-based array processing

Speech enhancement using CSS-based array processing

引用

International Conference on Acoustics, Speech, and Signal Processing (ICASSP)

作者： F. Asano S. Hayamizu Electro Technical Laboratory Tsukuba Japan

A method for recovering the LPC spectrum from a microphone array input signal corrupted by ambient noise is proposed. This method is based on the CSS (coherent subspace) method, which is designed for DOA (direction of arrival) estimation of broadband array input signals. The noise energy is reduced in the subspace domain by the maximum likelihood method. To enhance the performance of noise reduction, elimination of the noise-dominant subspace using projection is further employed, which is effective when the SNR is low and classification of noise and signals in the subspace domain is difficult. The results of the simulation show that some small formants, which cannot be estimated by the conventional delay-and-sum beamformer, were well estimated by the proposed method.

关键词： Speech enhancement Array signal processing Noise reduction Delay estimation Microphone arrays Direction of arrival estimation linear predictive coding Cascading style sheets Design methodology Signal design

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