Deepen Sinha - Academia.edu (original) (raw)
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California Polytechnic State University at San Luis Obispo
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Papers by Deepen Sinha
IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2005.
1 The AES has launched a new opportunity to recognize student members who author technical papers... more 1 The AES has launched a new opportunity to recognize student members who author technical papers. The Student Paper Award Competition is based on the preprint manuscripts accepted for the AES convention. Forty-two student-authored papers were nominated. The excellent quality of the submissions has made the selection process both challenging and exhilarating. The award-winning student paper will be honored during the convention, and the student-authored manuscript will be published in a timely manner in the Journal of the Audio Engineering Society. Nominees for the Student Paper Award were required to meet the following qualifications:
Journal of The Audio Engineering Society, 2012
ICC 2001. IEEE International Conference on Communications. Conference Record (Cat. No.01CH37240), 2001
Hybrid in band on channel (IBOC) digital audio broadcasting (DAB) simultaneously with analog ampl... more Hybrid in band on channel (IBOC) digital audio broadcasting (DAB) simultaneously with analog amplitude modulation (AM) has been proposed as a hybrid solution to digital audio broadcasting in the AM band. Adding digital transmission in the crowded AM band is a challenging proposition. To achieve FM like audio quality, an audio coder rate of 32-64 kbit/s may be required. One
Advances in Speech Coding, 1991
An important goal in current speech coding research is providing high-quality speech at low bit r... more An important goal in current speech coding research is providing high-quality speech at low bit rates (4.8–16 Kbps). Several methods [1]–[3] have been proposed recently to achieve this end. Compared to the conventional linear predictive (LP) vocoder [4], these methods employ an enhanced speech production model to synthesize speech. For example, instead of a single stage, the modulation filter now typically consists of two stages: i) a short-delay filter modeling the spectral envelope of speech, and ii) a long-delay filter modeling the spectral fine structure. Both are time-varying, all-pole filters and are derived from the original speech through LP analysis. Also, some information is provided about the excitation signal, which is selected by means of an analysis-by-synthesis procedure whereby a perceptually weighted error criterion is minimized In the multi-pulse linear predictive coder (MPLPC) [1], the excitation signal is a sequence of appropriately located and scaled impulses. In the code excited linear predictive coder (CELPC) [2], it is an entry from a codebook of white, gaussian noise sequences. In the self excited vocoder (SEV) [3], it is selected from the past history of the source excitation. As a result of these improvements, the above coders are able to synthesize high-quality speech at low bit rates.
IEEE Transactions on Information Theory, 1992
Abstract Two techniques for finding the discrete orthogonal wavelet of support less than or equal... more Abstract Two techniques for finding the discrete orthogonal wavelet of support less than or equal to some given integer that leads to the best approximation to a given finite support signal up to a desired scale are presented. The techniques are based on optimizing ...
IEEE Transactions on Broadcasting, 1999
IEEE Transactions on Broadcasting, 2005
IEEE Transactions on Speech and Audio Processing, 1994
A new scheme for coding speech at low bit rates (4.8-16 kb/s) but still maintaining high quality ... more A new scheme for coding speech at low bit rates (4.8-16 kb/s) but still maintaining high quality is described. Speech is regarded as a piecewise-stationary random signal and its synthesis is accomplished by means of a Kalman estimator at the decoder. The Kalman estimator requires for its operation a signal model and a sequence of measurements of the states of the model. A two-stage, time-varying, all-pole filter excited by white noise is used as the speech signal model. Linear combinations of speech samples taken at sparse but periodic intervals and provided in the form of innovations serve as measurements. The role of the encoder in the proposed scheme is seen as that of extracting the signal model parameters as well as forming the measurements and transmitting this information to the decoder. An optimum measurement strategy is developed for the estimator. A procedure for shaping the error spectrum of the synthesized speech is also described. Simulation studies show that coders based on the proposed scheme can provide high-quality speech at low bit rates. Important implementation details of such coders as well as their performance results for different choices of coder parameters are given. >
IEEE Workshop on Applications of Signal Processing to Audio and Acoustics, 2005.
1 The AES has launched a new opportunity to recognize student members who author technical papers... more 1 The AES has launched a new opportunity to recognize student members who author technical papers. The Student Paper Award Competition is based on the preprint manuscripts accepted for the AES convention. Forty-two student-authored papers were nominated. The excellent quality of the submissions has made the selection process both challenging and exhilarating. The award-winning student paper will be honored during the convention, and the student-authored manuscript will be published in a timely manner in the Journal of the Audio Engineering Society. Nominees for the Student Paper Award were required to meet the following qualifications:
Journal of The Audio Engineering Society, 2012
ICC 2001. IEEE International Conference on Communications. Conference Record (Cat. No.01CH37240), 2001
Hybrid in band on channel (IBOC) digital audio broadcasting (DAB) simultaneously with analog ampl... more Hybrid in band on channel (IBOC) digital audio broadcasting (DAB) simultaneously with analog amplitude modulation (AM) has been proposed as a hybrid solution to digital audio broadcasting in the AM band. Adding digital transmission in the crowded AM band is a challenging proposition. To achieve FM like audio quality, an audio coder rate of 32-64 kbit/s may be required. One
Advances in Speech Coding, 1991
An important goal in current speech coding research is providing high-quality speech at low bit r... more An important goal in current speech coding research is providing high-quality speech at low bit rates (4.8–16 Kbps). Several methods [1]–[3] have been proposed recently to achieve this end. Compared to the conventional linear predictive (LP) vocoder [4], these methods employ an enhanced speech production model to synthesize speech. For example, instead of a single stage, the modulation filter now typically consists of two stages: i) a short-delay filter modeling the spectral envelope of speech, and ii) a long-delay filter modeling the spectral fine structure. Both are time-varying, all-pole filters and are derived from the original speech through LP analysis. Also, some information is provided about the excitation signal, which is selected by means of an analysis-by-synthesis procedure whereby a perceptually weighted error criterion is minimized In the multi-pulse linear predictive coder (MPLPC) [1], the excitation signal is a sequence of appropriately located and scaled impulses. In the code excited linear predictive coder (CELPC) [2], it is an entry from a codebook of white, gaussian noise sequences. In the self excited vocoder (SEV) [3], it is selected from the past history of the source excitation. As a result of these improvements, the above coders are able to synthesize high-quality speech at low bit rates.
IEEE Transactions on Information Theory, 1992
Abstract Two techniques for finding the discrete orthogonal wavelet of support less than or equal... more Abstract Two techniques for finding the discrete orthogonal wavelet of support less than or equal to some given integer that leads to the best approximation to a given finite support signal up to a desired scale are presented. The techniques are based on optimizing ...
IEEE Transactions on Broadcasting, 1999
IEEE Transactions on Broadcasting, 2005
IEEE Transactions on Speech and Audio Processing, 1994
A new scheme for coding speech at low bit rates (4.8-16 kb/s) but still maintaining high quality ... more A new scheme for coding speech at low bit rates (4.8-16 kb/s) but still maintaining high quality is described. Speech is regarded as a piecewise-stationary random signal and its synthesis is accomplished by means of a Kalman estimator at the decoder. The Kalman estimator requires for its operation a signal model and a sequence of measurements of the states of the model. A two-stage, time-varying, all-pole filter excited by white noise is used as the speech signal model. Linear combinations of speech samples taken at sparse but periodic intervals and provided in the form of innovations serve as measurements. The role of the encoder in the proposed scheme is seen as that of extracting the signal model parameters as well as forming the measurements and transmitting this information to the decoder. An optimum measurement strategy is developed for the estimator. A procedure for shaping the error spectrum of the synthesized speech is also described. Simulation studies show that coders based on the proposed scheme can provide high-quality speech at low bit rates. Important implementation details of such coders as well as their performance results for different choices of coder parameters are given. >