Mark A. Jasiuk
Motorola
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Featured researches published by Mark A. Jasiuk.
international conference on acoustics, speech, and signal processing | 1990
Ira Alan Gerson; Mark A. Jasiuk
The vector sum excited linear prediction speech coder is presented. It utilizes a codebook with a structure that allows for a very efficient search procedure. Other advantages of the VSELP codebook structure are discussed, and a detailed description of an 8-kb/s VSELP coder is given. This coder was selected by the Telecommunications Industry Association (TIA) as the standard for use in North American digital cellular telephone systems. The coder uses two VSELP excitation codebooks, a gain quantizer which is robust to channel errors, and a novel adaptive pre/postfilter arrangement.<<ETX>>
Archive | 1991
Ira Alan Gerson; Mark A. Jasiuk
Vector Sum Excited Linear Prediction falls into the class of speech coders known as Code Excited Linear Prediction (CELP) (also called Vector Excited or Stochastically Excited) [1,4,5]. The VSELP speech coder was designed to achieve the highest possible speech quality with reasonable computational complexity while providing robustness to channel errors. These goals are essential for wide acceptance of low data rate (4.8-8 kbps) speech coding for telecommunications applications.
IEEE Journal on Selected Areas in Communications | 1992
Ira Alan Gerson; Mark A. Jasiuk
Techniques for improving the performance of CELP (code excited linear prediction)-type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function, which enhances the perceptual quality of the processed speech, is introduced. The combination of harmonic noise weighting and subsample pitch lag resolution significantly improves the coder performance for voiced speech. Strategies for reducing the speech coders data rate, while maintaining speech quality, are presented. These include a method for efficient encoding of the long-term predictor lags, utilization of multiple gain vector quantizers, and a multimode definition of the speech coder frame. A 5.9-kb/s VSELP speech coder that incorporates these features is described. Complexity reduction techniques which allow the coder to be implemented using a single fixed-point DSP (digital signal processor) are discussed. >
Journal of the Acoustical Society of America | 1998
Ira Alan Gerson; Mark A. Jasiuk
In a speech coder (100), excitation source gain information (802) is transmitted along with a coding mode indicator. The coding mode indicator indicates how the gain information is to be interpreted. In one embodiment, the coding mode indicator can also be utilized to control which of a plurality of excitation sources (202, 206-208) are utilized when synthesizing the speech. The coding mode itself is selected as a function of the periodicity of an input speech signal.
Journal of the Acoustical Society of America | 1998
Ira Alan Gerson; Mark A. Jasiuk; Matthew A Hartman
A Vector-Sum Excited Linear Predictive Coding (VSELP) speech coder provides improved quality and reduced complexity over a typical speech coder. VSELP uses a codebook which has a predefined structure such that the computations required for the codebook search process can be significantly reduced. This VSELP speech coder uses single or multi-segment vector quantizer of the reflection coefficients based on a Fixed-Point-Lattice-Technique (FLAT). Additionally, this speech coder uses a pre-quantizer to reduce the vector codebook search complexity and a high-resolution scalar quantizer to reduce the amount of memory needed to store the reflection coefficient vector codebooks. Resulting in a high quality speech coder with reduced computations and storage requirements.
international conference on acoustics, speech, and signal processing | 1991
Ira Alan Gerson; Mark A. Jasiuk
Techniques for improving the performance of CELP (code excited linear prediction) type speech coders while maintaining reasonable computational complexity are explored. A harmonic noise weighting function which enhances the perceptual quality of the processed speech is introduced. The combination of harmonic noise weighting and subsample resolution pitch significantly improves the coder performance for voiced speech. A 6.9 kb/s VSELP speech coder which incorporates subsample resolution pitch and harmonic noise weighting is described. Complexity reduction techniques are discussed which allow the coder to be implemented using a single fixed point digital signal processor.<<ETX>>
ieee workshop on speech coding for telecommunications | 1993
Ira Alan Gerson; Mark A. Jasiuk
The bit rate for the speech and channel coder for the half-rate GSM channel is 11400 bps. A 5600 bps VSELF speech coder designed for this application is described. This speech coder is one of two candidates being considered by GSM channel for the half-rate channel. It employs a novel strategy for vector quantization of the reflection coefficients (rj), which combines high coding efficiency, low codebook search complexity, and low storage requirements. A computationally streamlined version of a zero-pole spectral noise weighting function is implemented. An adaptive pitch prefilter and an adaptive spectral postfilter are used to improve the speech coders performance for both the tandemed and non tandemed cases.
Journal of the Acoustical Society of America | 1994
Ira Alan Gerson; Mark A. Jasiuk
A speech encoder uses a soft interpolation decision for spectral parameters. For each frame, the encoder first calculates the residual energy for interpolated spectral parameters, and then calculates the residual energy for non-interpolated spectral parameters. The encoder then compares these residual energy calculations. If the encoder determines that the interpolated spectral parameters yields the lowest residual energy, it indicates to a far-end decoder to use the interpolated values for the current frame. Otherwise, it indicates to the far-end decoder to use the non-interpolated values for the current frame. The encoder signals the far-end decoder as to which spectral parameters (interpolated or non-interpolated values) to use by encoding and transmitting a special signalling bit.
Journal of the Acoustical Society of America | 1997
Ira Alan Gerson; Mark A. Jasiuk
A digital speech coder utilizes harmonic noise weighting to overcome some limitations of low-rate CELP-type speech coders in reproducing voiced speech. In addition to a short term correction factor, which constitutes spectral noise weighting as known in the art, a long term pitch correction factor is utilized to provide harmonic noise weighting. The inclusion of harmonic noise weighting in a speech coder more efficiently utilizes noise-masking properties of a speech signal, allowing synthesis of a higher quality speech at a given bit rate.
international conference on acoustics, speech, and signal processing | 2005
Udar Mittal; James P. Ashley; Edgardo M. Cruz-Zeno; Mark A. Jasiuk
Codebook searches in analysis-by-synthesis speech coders typically involve minimization of a perceptually weighted squared error signal. Minimization of the error over multiple codebooks is often done in a sequential manner, resulting in the choice of overall excitation parameters being sub-optimal. In this paper, we propose a joint excitation parameter optimization framework in which the associated complexity is slightly greater than the traditional sequential optimization, but with significant quality improvement. Moreover, the framework allows joint optimization to be easily incorporated into existing pulse codebook systems with little or no impact on the codebook search algorithms. This technique is part of the 3GPP2 source-controlled variable-rate multimode wideband speech codec (VMR-WB) rate set 1 standard.