A Study on Pulse Search Algorithms for Multipulse Excited Speech Coder Realization

Ozawa, K.; Ono, Shigeru; Araseki, T.

doi:10.1109/jsac.1986.1146292

Cited by 15 publications

(5 citation statements)

References 11 publications

Supporting

Mentioning

Contrasting

Unclassified

Order By: Relevance

“…Prom these in vestigations, it was observed th a t when the vector dimension exceeds 16 samples, a high computational complexity is incurred during the search for the optimal excitation sequence. In such cases, sub-optimal fast searches can be employed to increase the ef ficiency of the algorithm [115]. When the length of the excitation vector is longer than 12 samples, the results exhibit a considerable drop in the S N R seG' Both measures indicate th at the optimal dimension for the excitation sequence is 12 samples with an update interval of 24 samples.…”

Section: 23 C H Oice O F U P D a Te In Terval And E X Cita Tio N V Ec...mentioning

confidence: 99%

Combined speech and audio coding with bit rate and bandwidth scalability

Farrugia¹

2001

7th European Conference on Speech Communication and Technology (Eurospeech 2001)

View full text Add to dashboard Cite

The past two decades have witnessed a rapid expansion within the telecommunications industry. This growth has been primarily motivated by the proliferation of digital communication systems and services which have become easily available through wired and wireless systems. Current research trends involve the integration of speech, audio, video and d ata channels into true multimedia communications over fixed and mobile networks. However, while the available bandw idth in wired terrestrial networks is rela tively cheap and expandable, it becomes a limited resource in satellite and cellular-radio systems. In order to accommodate an ever growing number of users while maintaining high quality and low operational costs, it is necessary to maximise spectral efficiency. This has given rise to the development of high rate compression techniques w ith the ability to adapt to a broad class of input signals and to varying network resources.The research carried out in this thesis has mainly focused on the design of a single algorithm for compressing speech and audio signals sampled at different rates. The algorithms are based on the analysis-by-synthesis linear prediction coding (AbS-LPC) scheme, which has been widely employed in various speech coding standards. However, this bit rate reduction technique is based on the speech production mechanism and as such provides a rigid structure which presents a major lim itation for audio coding. In order to improve the audio quality at low rates and to compensate for the errors incurred by the linear prediction during segments of high transitions, the algorithms employ an efficient pulse excitation structure which represents the short innovation sequences with sparse unit magnitude pulses. The scheme proposed for the compression of telephone bandw idth speech and audio signals at 12kb/s achieves similar quality to the G.728 coder at 16kb/s and higher audio quality than the GSM-EFR standard at 12.2kb/s. W ideband speech and audio coding schemes have been designed using both the fullband approach at bit rates of 17 and 19kb/s and also the split band technique at a bit rate of 20kb/s. The perceptual quality is comparable to the G.722 coder operating at 48kb/s.The subband decomposition technique is also adapted to code speech and audio signals sampled at 32kHz. The quality of the coder at 28kb/s is similar to the quality achieved by the MP3 coder at 32kb/s. The algorithm also provides bandw idth and bit rate scalability ranging from 12 to 64kb/s, making it ideal for deployment in rate-adaptive communication systems. A ck n ow led gem en ts I would like to take this opportunity to express my gratitude to Prof. Ahmet Kondoz, my PhD project supervisor, whose guidance and suggestions throughout my research were most helpful and much appreciated. I am also indebted to all my colleagues in the M ultimedia Communications group, especially Stéphane Villette and Yong Duk Cho, whose suggestions have been invaluable. I would like thank my family, especially my grandmother, for the constant support,...

show abstract

Section: 23 C H Oice O F U P D a Te In Terval And E X Cita Tio N V Ec...mentioning

confidence: 99%

Combined speech and audio coding with bit rate and bandwidth scalability

Farrugia¹

2001

7th European Conference on Speech Communication and Technology (Eurospeech 2001)

View full text Add to dashboard Cite

show abstract

“…Not surprisingly, similar algorithms have been developed for subset selection in other application contexts. For instance, the matching pursuit algorithm was developed independently for speech coding in the context of pulse location determination in multipulse speech coders [30,33,34].…”

Section: Basic Matching Pursuit (Bmp)mentioning

confidence: 99%

“…Motivated by applications, we consider the computational complexity of the algorithms in two contexts: one where the dictionary D is variable (time dependent), and the other where the dictionary D is fixed (time independent). For instance, in multipulse speech coding the dictionary varies from frame to frame [12,30,33,34] giving rise to the variable dictionary scenario, while a fixed dictionary can be used in time-frequency representations of a signal [4,25]. The choice of a fixed or variable dictionary is important because it involves a trade-off between memory usage and computation.…”

Section: Computation Analysismentioning

confidence: 99%

Forward sequential algorithms for best basis selection

Cotter

Adler

Rao

et al. 1999

IEE Proc., Vis. Image Process.

121

View full text Add to dashboard Cite

Recently, the problem of signal representation in terms of basis vectors from a large, "overcomplete", spanning dictionary has been the focus of much research. Achieving a succinct, or "sparse", representation is known as the problem of best basis representation. We consider methods which seek to solve this problem by sequentially building up a basis set for the signal. Three distinct algorithm types have appeared in the literature which we term Basic Matching Pursuit (BMP), Order Recursive Matching Pursuit (ORMP) and Modified Matching Pursuit (MMP). The algorithms are first described and then their computation is closely examined. Modifications are made to each of the procedures which improve their computational efficiency. Each algorithm's complexity is considered in two contexts: one where the dictionary is variable (time dependent), and the other where the dictionary is fixed (time independent). Experimental results are presented which demonstrate that the ORMP method is the best procedure in terms of its ability to give the most compact signal representation, followed by

show abstract

“…1. In this method, excitation signal is represented by multi-pulse [6], [7]. The amplitudes or signs of all multipulse in a sub-frame are simultaneously vector quantized to improve the quantization performance [2], [3].…”

Section: Overview Of Multi-pulse Based Celpmentioning

confidence: 99%

4 kb/s multi-pulse based CELP speech coding using excitation switching

Ozawa

1999

1999 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings. ICASSP99 (Cat. No.99CH36258)

Self Cite

View full text Add to dashboard Cite

This paper proposes an MP-CELP (Multi-Pulse-based CELP)speech coding at 4 kb/s. In MP-CELP, amplitudes or signs of multi-pulse excitation are simultaneously vector quantized (VQ). In order to improve speech quality for background noise conditions, excitation signal is switched between voiced and unvoiced speech, and the number of pulse is greatly increased for unvoiced speech by restricting pulse locations. Further, in order to improve voiced speech quality, the optimal combination among adaptive codebook lag, pulse location, sign codevector and gain codevector is selected which minimizes distortion by employing delayeddecision search. The subjective evaluation results show that speech quality for 4 kb/s MP-CELP is close to that for ITU-T G.723.1 (6.3 kb/s) and G.729 (8 kbls) in M-IRS clean speech condition. For background noise conditions, the introduction for the excitation switching and the pulse location restriction significantly improves MOS value by 0.4. However, further improvement is still required, except for interference talker condition.

show abstract

A Study on Pulse Search Algorithms for Multipulse Excited Speech Coder Realization

Cited by 15 publications

References 11 publications

Combined speech and audio coding with bit rate and bandwidth scalability

Combined speech and audio coding with bit rate and bandwidth scalability

Forward sequential algorithms for best basis selection

4 kb/s multi-pulse based CELP speech coding using excitation switching

Contact Info

Product

Resources

About