An adaptive multi rate wideband speech codec with adaptive gain re-quantization

Erdmann, C.; Vary, Peter; Fischer, Katrin; Stegmann, Joachim; Quinquis, Catherine; Massaloux, D.; Kovesi, B.

doi:10.1109/scft.2000.878431

Cited by 3 publications

(5 citation statements)

References 7 publications

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“…In this paper, we presented the salient features of AMR-WB codec and investigated the possibilities to further improve performance for GSM system. One of the techniques we recommended is the SB-CELP algorithm that makes the use of adaptive gain re-quantization of fixed codebook [10]. The other recommendation to incorporate is the use of adaptive thresholds for various codec modes as per the current speech quality [11].…”

Section: Discussionmentioning

confidence: 99%

“…Due to this reason, there are continuous efforts to improve the speech quality further. The quality of speech may be improved by adaptive gain re-quantization of fixed codebook (FCB) [10]. This approach is called Sub-band CELP (SB-CELP).…”

Section: Quality Of Sspechmentioning

confidence: 99%

“…The various bit-rates are integrated in a common structure where the scalability is realized by exchanging fixed excitation codebook while leaving other codec parameters invariant. The upper SB is coded using a very low bit-rate representation based on bandwidth extension technique [10].…”

Section: Quality Of Sspechmentioning

confidence: 99%

“…The SB-CELP encoder used a method to determine the FCB gain factor that helps to overcome the unwanted gain fluctuations as they appear in CELP coders when coding pure background noise segment or speech in background noise [10].…”

Section: Quality Of Sspechmentioning

confidence: 99%

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An Investigation into the Performance Enhancement and Salient Features of AMR-WB Speech Codec Used in GSM Communication

Ahmad¹,

Beg²

2012

IOSRJECE

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Section: Discussionmentioning

confidence: 99%

Section: Quality Of Sspechmentioning

confidence: 99%

Section: Quality Of Sspechmentioning

confidence: 99%

Section: Quality Of Sspechmentioning

confidence: 99%

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An Investigation into the Performance Enhancement and Salient Features of AMR-WB Speech Codec Used in GSM Communication

Ahmad¹,

Beg²

2012

IOSRJECE

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“…A diagram illustrating the structure of the wideband coder is shown in nique is adopted by various coders, some dividing the spectrum into unequal subbands [130,121,131]. Some coders even try to model the critical bands of the human auditory system [132] and split the spectrum accordingly.…”

Section: 2 System Overviewmentioning

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)

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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,...

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A candidate proposal for a 3GPP adaptive multi-rate wideband speech codec

Erdmann

Vary²,

Fischer³

et al.

2001 IEEE International Conference on Acoustics, Speech, and Signal Processing. Proceedings (Cat. No.01CH37221)

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This paper describes an adaptive multi-rate wideband (AMR-WB) speech codcc proposcd for the GSM system and also for the evolving Third Generation (3G) mobile speech services. The speech codec is based on SB-CELP (Splitband-Code-Excited Linear Prediction) with five modes operating bit rates from 24kbit/s down to S.lkbit/s. The respective channel coding schemes are based on RSC (Recursive Systematic Code) and UEP (Unequal Error Protection). Both, source and channel codec are designed as homogenous as possible to guarantee robust transmission on current and future mobile radio channels.

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An adaptive multi rate wideband speech codec with adaptive gain re-quantization

Cited by 3 publications

References 7 publications

An Investigation into the Performance Enhancement and Salient Features of AMR-WB Speech Codec Used in GSM Communication

An Investigation into the Performance Enhancement and Salient Features of AMR-WB Speech Codec Used in GSM Communication

Combined speech and audio coding with bit rate and bandwidth scalability

A candidate proposal for a 3GPP adaptive multi-rate wideband speech codec

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