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Enhanced Variable Rate Codec

About: Enhanced Variable Rate Codec is a research topic. Over the lifetime, 357 publications have been published within this topic receiving 4842 citations. The topic is also known as: EVRC.


Papers
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Proceedings ArticleDOI
28 May 2015
TL;DR: This work investigates the different factors like bit rate, algorithmic delay, implementation, and more importantly robustness of codec's perceptual quality to noise that play a vital role in choosing a speech codec and evaluating its performance.
Abstract: With wireless acoustic sensor network extending to the services like surveillance of sensitive areas, such as Line of Control, or unmanned terrains, interest in robust, narrowband and low bit rate speech codecs is increasing. This has resulted in a need for evaluation of such codecs. This work investigates the different factors like bit rate, algorithmic delay, implementation, and more importantly robustness of codec's perceptual quality to noise. These factors play a vital role in choosing a speech codec and evaluating its performance. This work examines the selected narrow band speech codecs using ITU-T (International Telecommunication Union) recommendation P.862 tool. This study can also help to choose a speech codec in myriad applications like Wireless Acoustic Sensor Networks, Cellular Telephony, VoIP, Underwater Acoustic Monitoring and many other applications where bandwidth plays a vital role and data compression is inevitable.

2 citations

Book ChapterDOI
13 Aug 2007

1 citations

Proceedings ArticleDOI
08 Dec 2008
TL;DR: This paper describes an embedded speech and audio codec which is based on ITU-T Recommendation G.722.1; it can process 7 kHz bandwidthspeech and audio signal at scalable bit rates and adds two modules: the energy ordering of sub-band and the processing of bit-stream truncation.
Abstract: This paper describes an embedded speech and audio codec which is based on ITU-T Recommendation G.722.1; it can process 7 kHz bandwidth speech and audio signal at scalable bit rates. Based on the G.722.1 of ITU-T, this algorithm adds two modules: the energy ordering of sub-band and the processing of bit-stream truncation. Furthermore, it does some modification on the categorization and noise-fill modules. It makes sure that the codec could produce embedded bit-stream, so this codec had more robustness in the transmission. The test results by ITU-T PESQ show that this codec has good performance as G.722.1 at the same bit-rates.

1 citations

Journal Article
TL;DR: This paper describes and implements wideband speech codec algorithms on the DSP56156 digital signal processor and describes the method proposed by the British Telecom Research Laboratory in the GSM Pan-European study on digital cellular land mobile radio.
Abstract: This paper describes and implements wideband speech codec algorithms on the DSP56156 digital signal processor. Two wideband speech codec algorithms are described. The first algorithm is based on the G722 recommendation provided by the International Telegraph and Telephone Consultative Committee Study Group XVIII for 7 kHz audiocoding at 64 kbits/s. To reduce the bit rate and improve the speech quality, a second algorithm is proposed for 7 kHz audiocoding at 40 kbits/s. The second algorithm is based on the method proposed by the British Telecom Research Laboratory in the GSM Pan-European study on digital cellular land mobile radio. The results obtained for the two wideband codec algorithms are included.

1 citations

11 Oct 2001

1 citations


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Performance
Metrics
No. of papers in the topic in previous years
YearPapers
20176
20167
201513
20149
201311
20128