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An Algorithm for Simple Differential Speech Coding Based on Backward Adaptation Technique
2018
Informatica
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This paper presents a simple differential speech signal coding algorithm, based on backward adaptation. ...
Experimental results show that our simple differential speech coding algorithm satisfies the G.712 Recommendation for high-quality speech coding at the bit rate of 6 bits per sample. ...
Conclusion This paper has presented a simple, differential wideband speech signal coding algorithm, with the implementation of backward adaptation technique. ...
doi:10.15388/informatica.2018.180
fatcat:ud2hacxkgze4nkhsncq6dyhuvi
Wide-Band Speech Coding Using Kernel Methods And Bandwidth Extension Based On Parametric Stereo
2012
Zenodo
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The core part of the BWE technique is an efficient coding scheme for the narrow-band speech. Adaptive differential pulse code modulation (ADPCM) is a technique vastly used for this purpose [7] . ...
The utilization of kernel methods, in the framework of the backward ADPCM technique, for narrow-band speech coding is studied for the first time. ...
doi:10.5281/zenodo.52429
fatcat:dcubvfhv5fen3fgaiee4od2r4a
A low delay 16 kb/s speech coder
1991
IEEE Transactions on Signal Processing
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IYENGAR AND KABAL: LOW DELAY 16 KB/S SPEECH CODER 1053 ...
uevd. y J L I I~~~~ ~I~\ W I Y U W~~ yu6u u u u p o \LLIJI r~p r v l u g~ I U I 1lllJ I L L L IKH.YD,ALIlWlY3, llhC LIIUSC JUUlIliilS UI ULllrr prulessiuflal societies, is not a necessary prerequisite for ...
This coding algorithm utilizes backward adaptation schemes for the predictor and quantizer, i.e., the adaptations are based on an analysis of the past quantized data. ...
doi:10.1109/78.80962
fatcat:abs45jwzrbbediejm3pfdq7r7y
Low Complex Forward Adaptive Loss Compression Algorithm and Its Application in Speech Coding
2011
Journal of Electrical Engineering
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Low Complex Forward Adaptive Loss Compression Algorithm and Its Application in Speech Coding This paper proposes a low complex forward adaptive loss compression algorithm that works on the frame by frame ...
Accordingly, we can reasonably believe that our algorithm will find its practical implementation in the high quality coding of signals, represented with less than 8 bits/sample, which as well as speech ...
The conceptual difference between the adaptive coding algorithms is based on the manner in which the adaptation is performed -whether it is performed forward, ie, from the input sequence or backward, ie ...
doi:10.2478/v10187-011-0003-5
fatcat:74ocedjgxfavfaqbux3k6w7bhi
Backward-adaptive lossless compression of video sequences
2002
IEEE International Conference on Acoustics Speech and Signal Processing
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Key features of the coder are a pixel-neighborhood backward-adaptive temporal predictor, an intra-frame spatial predictor and a differential coding scheme of the spectral components. ...
We present our new low-complexity compression algorithm for lossless coding of video sequences. ...
Key features of the coder are a pixel-neighborhood backward-adaptive temporal predictor, an intra-frame spatial predictor and a differential coding scheme of the spectral components. ...
doi:10.1109/icassp.2002.5745388
dblp:conf/icassp/CarottiMM02
fatcat:2beqvvoexnbhjj2fhoq3rtpmqe
Backward-adaptive lossless compression of video sequences
2002
IIEEE International Conference on Acoustics Speech and Signal Processing
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Key features of the coder are a pixel-neighborhood backward-adaptive temporal predictor, an intra-frame spatial predictor and a differential coding scheme of the spectral components. ...
We present our new low-complexity compression algorithm for lossless coding of video sequences. ...
Key features of the coder are a pixel-neighborhood backward-adaptive temporal predictor, an intra-frame spatial predictor and a differential coding scheme of the spectral components. ...
doi:10.1109/icassp.2002.1004646
fatcat:w3qc5har2vdhjafl3huiqf5dom
Model-based multirate representation of speech signals and its application to recovery of missing speech packets
1997
IEEE Transactions on Speech and Audio Processing
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Next, a new sample-interpolation algorithm based on the multirate Kalman reconstruction filter is proposed to reduce speech quality degradation caused by packet losses. ...
Subjective tests indicate that the proposed Kalman-based sample-interpolation algorithm performs better than the conventional odd-even sample-interpolation procedure for mitigating the effects of random ...
ACKNOWLEDGMENT The authors thank the reviewers for their constructive comments and suggestions, which have greatly improved the quality of this manuscript. ...
doi:10.1109/89.568729
fatcat:ozo5lvxwajcm5adolpglr6ez64
mLEARn: An Implementation of Multi-layer Perceptron in C++
2021
Journal of Open Source Education
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The techniques and algorithms implemented represent existing approaches in machine learning. mLEARn is written using simple C++ constructs. ...
The aim of mLE ARn is to provide a simple and extendable machine learning platform for students in courses involving C++ and machine learning. ...
The author would like to acknowledge support for the speech and language technologies program. ...
doi:10.21105/jose.00059
fatcat:6fdnucbm7jg7lhigqzuvvatsn4
Speech Technology Progress Based on New Machine Learning Paradigm
2019
Computational Intelligence and Neuroscience
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Speech technologies have been developed for decades as a typical signal processing area, while the last decade has brought a huge progress based on new machine learning paradigms. ...
This review article on speech signal analysis and processing, corresponding machine learning algorithms, and applied computational intelligence aims to give an insight into several fields, covering speech ...
work described in this paper was supported in part by the Ministry of Education, Science and Technological Development of the Republic of Serbia, through the project "Development of Dialogue Systems for ...
doi:10.1155/2019/4368036
pmid:31341467
pmcid:PMC6614991
fatcat:yfwrwisz7jgrtlijfpj7rkuuoi
Speech coding: a tutorial review
1994
Proceedings of the IEEE
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The objective of this paper is to provide a tutorial overview of speech coding methodologies with emphasis on those algorithms that are part of the recent low-rate standards f o r cellular communications ...
Most of these coders incorporate mechanisms to: represent the spectral properties of speech, provide f o r speech waveform matching, and "optimize" the coder's performance for the human ear. ...
There are two types of A- VQ, namely, forward-adaptive and backward-adaptive. In backward-adaptive VQ, codebook updating is based on past data which are also available at the decoder. ...
doi:10.1109/5.326413
fatcat:6mvwz3wngjbcbi2h2bypig3na4
State of the art and trends in speech coding
1995
Philips Journal of Research
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An introductory review of some basic speech coding techniques covers the most important properties of speech production and hearing, the ubiquitous techniques of quantization and linear prediction, and ...
The paper, which primarily deals with narrowband speech coding systems, is concluded by a review of the state of affairs and an outline of the future trends in the area of wideband speech coding. ...
For the sake of comparison, the SNR of an 8-bit backward adaptive quantizer, again for a sinusoidal input signal, is also shown in Fig. 4 . ...
doi:10.1016/0165-5817(96)81591-9
fatcat:lzgivkduc5hfrhj752ucmteaim
REAL-TIME SPEECH COMPRESSION BY USING CODE EXCITED LINEAR PREDICTION ALGORITHM
2009
International Journal on Intelligent Electronic Systems
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Speech compression is proposed based on code excited linear prediction algorithm and implementation in DSP algorithm. ...
Algorithm based on three-stage technique which involves simulate, evaluate, debug and implementation in G.723 low delay code excited linear prediction (LD-CELP)[4] algorithm. ...
The description of the speech-coding algorithm of this Codec is made in terms of bit-exact, fixed-point mathematical operations
B Encoder principle The coder is based on the principles of linear prediction ...
doi:10.18000/ijies.30045
fatcat:6drkvameonb3xosvdfixmedz4q
Bandwidth-efficient wireless multimedia communications
1998
Proceedings of the IEEE
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We then summarize the fundamental concepts of modulation, introduce an adaptive modem scheme, and argue that third-generation transceivers might become adaptively reconfigurable under network control in ...
for wireless communications are addressed. ...
Such a simple video codec schematic based on simple frame differencing is shown in Fig. 6 . ...
doi:10.1109/5.681368
fatcat:5ycvcgjfk5bjbfblcqckuva37e
Silent and voiced/unvoiced/mixed excitation (four-way) classification of speech
1989
IEEE Transactions on Acoustics Speech and Signal Processing
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In an isolated word recognition system, the search to identify a test word among all possible candidates can be reduced by using such a simple coding technique. ...
An algorithm for recognizing voice onset and offset using the EGG is described in [ 1 11 and is an extension of the one in [3], [4] . ...
doi:10.1109/29.46561
fatcat:ddmd6xltxzcpdkg2shfdlb4wre
Perceptual coding of digital audio
2000
Proceedings of the IEEE
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In response to this need, considerable research has been devoted to the development of algorithms for perceptually transparent coding of high-fidelity (CD-quality) digital audio. ...
This paper reviews algorithms for perceptually transparent coding of CD-quality digital audio, including both research and standardization activities. This paper is organized as follows. ...
ACKNOWLEDGMENT The authors would like to thank the anonymous reviewers for their constructive comments and corrections. ...
doi:10.1109/5.842996
fatcat:jkfvoxg7zrcyxg6fyahe4u73pu
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