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检索条件"主题词=Linear Predictive coding"

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Nonlinear filtering of the LPC residual for noise suppression and speech quality enhancement

Nonlinear filtering of the LPC residual for noise suppressio...

引用

IEEE Workshop on Speech coding For Telecommunications

作者： P. Mermelstein Yasheng Qian INEtS-Télécommunications Université du Quàbec Canada

We study enhancing the quality of coded speech by nonlinear filtering the linear-prediction residual to make it more periodic. By attenuating the low-magnitude samples, more of the residual energy can be concentrated in the regularly spaced major excitation pulses. Wideband additive noise in the speech signal is attenuated because its residual energy is spread in time, rather than being concentrated in narrow temporal regions.

关键词： Filtering linear predictive coding Speech enhancement Speech synthesis Speech analysis Nonlinear filters Additive noise Distortion Signal synthesis Attenuation

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Spectral transformation for musical tones via time domain filtering

Spectral transformation for musical tones via time domain fi...

引用

IEEE Workshop on Applications of Signal Processing to Audio and Acoustics

作者： S. Yim Yinong Ding A. McCree DSP Research and Development Center Texas Instruments Inc. Dallas TX USA

We present a spectral transformation technique for musical tones. It can be used to modify the brightness or timbre of a musical tone. We perform spectral transformation by modifying the line spectral frequencies or LPC roots of the original spectral envelope and filtering the tone with a spectral transformation filter in the time domain. One of the application is pitch modification where frequency scaling is used to modify the fundamental frequency of a tone and spectral transformation is used to capture the original brightness and timbre. We have applied this spectral transformation technique to restore the original brightness of frequency scaled musical tones produced from a variety of musical instruments, such as piano, trumpet, violin, flute and bassoon by one octave up or down. In all experiments conducted, the pitch modified tones resemble the original brightness and timbre. This technique allows us to control the timbre of musical tones such as brightness, friction or other modulation by manipulating the spectral parameters.

关键词： Frequency Timbre Brightness Filtering linear predictive coding Spectral shape Acoustic distortion Instruments Filters Friction

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A design of transform coder for both speech and audio signals at 1 bit/sample

A design of transform coder for both speech and audio signal...

引用

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

作者： T. Moriya N. Iwakami A. Jin K. Ikeda S. Miki NTT Human Interface Laboratories Tokyo Japan

This paper proposes a speech and audio coder which operates at 1 bit/sample, namely an 8 kbit/s coder for 8 kHz sampling or a 16 kbit/s coder for 16 kHz sampling. The basic structure is inherited from a Twin VQ (transform domain weighted interleave vector quantization) high-quality audio coding scheme. A periodical component extraction scheme is newly added to the quantization of the MDCT coefficients. This scheme is found to be effective for reducing distortion and improving the robustness against channel errors. The qualities for music signals at 8 kbit/s are better than those of G.729 at the same bit rates, while they are worse for clean speech. The qualities at 16 kbit/s are comparable to or better than those of G.722 at 48 kbit/s.

关键词： Signal design linear predictive coding Discrete transforms Audio coding Multiple signal classification Frequency domain analysis Speech coding Delay Signal analysis Frequency estimation

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A lattice low-delay code-excited linear prediction speech coder at 16 kb/s

A lattice low-delay code-excited linear prediction speech co...

引用

IEEE Symposium on Computers and Communications (ISCC)

作者： S.H. El-Ramly R. El-Kouatly M.N. Saleh Electronics and Comm. Eng. Department Faculty of Engineering Ain Shams University Cairo Egypt

This paper presents a lattice low-delay code-excited linear prediction speech coder (LLD-CELP) based on the analysis-by-synthesis configuration. The coder achieves a one-way coding delay of less than 2 ms by making both the LPC predictor and the excitation gain backward-adaptive, and by using a small excitation vector size of four samples. The introduction of a lattice filter as a short term predictor, and a perceptual weighting filter has significant advantages, such as fast tracking of speech signal nonstationarities, simple stability verification, and uniform distribution of the computation load. The introduction of backward adaptive prediction gain imbedded in the excitation codebook achieves a lower complexity with the same quality and with the same bit rate.

关键词： Lattices Filters Speech synthesis Speech analysis Delay Vectors Stability Bit rate Decoding linear predictive coding

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Utterance dependent parametric warping for a talker-independent HMM-based recognizer

Utterance dependent parametric warping for a talker-independ...

引用

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

作者： D.J. Mashao J.E. Adcock LEMS Division of Engineering Brown University Providence RI USA

In an effort to improve the recognition performance of talker-independent speech systems, many adaptive methods have been proposed. The methods generally seek to exploit the higher recognition performance rate of talker-dependent systems and extend it to talker-independent systems. This is achieved by some form of placing talkers into several categories, usually using gender or vocal-tract size. We investigate a similar idea, but categorize each utterance independently. An utterance is processed using several spectral compressions, and the compression with the maximum likelihood is then used to train a better model. For testing, the spectral compression with the maximum likelihood is used to decode the utterance. While the spectral compressions divided the utterances well, this did not translate into significant improvement in performance, and the computational cost increase was significant.

关键词： Hidden Markov models Speech recognition Adaptive systems Testing Maximum likelihood decoding Computational efficiency Speech enhancement Shape Character recognition linear predictive coding

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On a time reduction of pitch searching by the regular pulse search technique in the CELP vocoder

On a time reduction of pitch searching by the regular pulse ...

引用

MILCOM, Military Communications Conference

作者： YoungHo Park JaeChan Yang SangMok Sohn MyungJin Bae R&D Center Seoul Mobile Telecom Seoul South Korea Department of Telecommunication Engineering Soongsil University Seoul South Korea

ISBN: (纸本)0780342496

Code excited linear prediction (CELP) speech coders exhibit good performance at data rates as low as 4800 bps. The major drawback of CELP type coders is the large amount of computational time. In this paper, we propose a new pitch codebook search method that restricts some range of the pitch searching by using the regular pulse excitation (RPE) searching technique. Applying the proposed method to the 8 kbps CELP vocoder, we can get approximately 48% complexity reduction in the pitch search.

关键词： Vocoders Speech analysis Speech synthesis Filters Signal synthesis Speech coding linear predictive coding Performance analysis Signal processing Research and development

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Audio coding using sinusoidal excitation representation

Audio coding using sinusoidal excitation representation

引用

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

作者： Wen-Whei Chang De-Yu Wang Li-Wei Wang Department of Communication Engineering National Chiao Tung university Hsinchu Taiwan

ISBN: (纸本)0818679190

Most LPC-based audio coders employ simplistic noise-shaping operations to perform psychoacoustic control of quantization noise. In this paper, we report on new approaches to exploiting perceptual masking in the design of adaptive quantization of LPC excitation parameters. Due to its localized spectral sensitivity, sinusoidal excitation representation is preferred to spectrally flat signals for use in excitation modeling. Simulation results indicate that the proposed multisinusoid excited coder can deliver high quality audio reproduction at the rate of 72 kb/s.

关键词： Audio coding linear predictive coding Quantization Psychoacoustic models Psychology Filters Frequency Noise shaping Bit rate Signal processing

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An experimental audio codec based on warped linear prediction of complex valued signals

An experimental audio codec based on warped linear predictio...

引用

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

作者： A. Harma U.K. Laine M. Karjalainen Laboratory of Acoustics and Audio Signal Processing Helsinki University of Technology Espoo Finland

Bark-scale warped linear prediction (WLP) is a very potential core for a monophonic perceptual audio codec. In the current paper the WLP scheme is extended for processing complex valued signals (CWLP). Three different methods of converting a stereo signal to one complex valued signal are introduced. The philosophy behind the coding scheme is to integrate some aspects of modern wideband audio coding (e.g. perceptuality and stereo signal processing) into one computational element in order to find a more holistic and economic way of processing.

关键词： Codecs linear predictive coding Signal processing Filters Wideband Audio coding Signal analysis Autocorrelation Vectors Symmetric matrices

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On a smoothing method by time-axis control with the zero-crossing of speech signal

On a smoothing method by time-axis control with the zero-cro...

引用

MILCOM, Military Communications Conference

作者： MyungJin Bae JongDeuk Kim Speech Communication Laboratory Department of Telecommunication Engineering Soongsil University Seoul South Korea

ISBN: (纸本)0780342496

In the area of speech synthesis techniques, waveform coding with high quality is mainly used for synthesis by analysis. In order to apply waveform coding, a pitch alteration technique is required in prosody control. The spectrum distortion of the pitch alteration method by time-axis control increases exponentially when the alteration rate is increased. In this paper, we propose a new pitch alteration method that uses a smoothing technique. This method changes the pitch period by partially controlling the time-axis of each zero crossing of the speech signal. This is a time domain method that preserves the phase components of the waveform and has little spectrum distortion of 4.3% for 70% pitch change.

关键词： Smoothing methods Speech synthesis linear predictive coding Time frequency analysis Speech analysis Signal synthesis Phase distortion Frequency domain analysis Filters Cepstrum

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Speech aid for the deaf based on a representation of the vocal tract: the vowel module

Speech aid for the deaf based on a representation of the voc...

引用

Annual International Conference of the IEEE Engineering in Medicine and Biology Society (EMBC)

作者： P.M.T. de Oliveira M.N. Souza Biomedical Engineering Program COPPE Federal University of Rio de Janeiro Brazil

Presents a new method for speech training of deaf and hard-of-hearing people, using a computer accompanied with an A/D sound card. The idea is to produce a visual feedback of the deaf person's voice, also giving information about what must be done in order to correctly produce the required speech sound. The visual representation attempts to reproduce the configuration of the speaker's vocal tract in real time, during vowel articulation. Since changes in this configuration reflect changes of the spectral envelope of the speech signal, we need to solve the inverse problem: obtaining the vocal tract shape from the speech signal. To solve this problem, signal processing techniques such as linear predictive coding were used. We hope that this representation can be more effective than other representations in use today, such as spectra and spectrograms, due to its closer association with the speech production processes.

关键词： Deafness linear predictive coding Speech processing Feedback Resonance Signal processing Tongue Mouth Data acquisition Biomedical engineering

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